From philq at qsystemsengineering.com  Wed Aug  1 00:02:35 2012
From: philq at qsystemsengineering.com (Phil Quesinberry)
Date: Tue, 31 Jul 2012 16:02:35 -0400
Subject: [Freeswitch-users] Setting effecting_caller_id_name
Message-ID: <048301cd6f57$70b8ce90$522a6bb0$@com>

One thought did occur to me.

Would 'caller_id_name' not be set to the CNAM here because of the fact that
the lookup has already been cached with memcache?  If that's the case, is
there a way around this behavior other than disabling memcache?

- Phil

_____________________________________________
From: Phil Quesinberry 
Sent: Tuesday, July 31, 2012 12:20 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: Setting effecting_caller_id_name


Ok, the problem here is that the variable caller_id_name contains the Caller
ID number instead of the CNAM that was looked up.  Is there a variable to
look at and change for the CNAM info?  Info shows the CNAM info in
Caller-Caller-ID-Name but attempts to match on variations of that have
failed.  Obviously I'm missing some basic piece of info here but I haven't
been able to find it, even within the FreeSwitch book.

I've pasted a small section of the relevant console output below.

I should also mention that I'm doing this check within public.xml since I
want it to apply to all incoming calls.

Thanks,

- Phil


2012-07-31 11:33:30.800571 [INFO] mod_dialplan_xml.c:485 Processing
4435551212 <4435551212>->4105551212 in context public
Dialplan: sofia/external/4435551212 at 140.239.xx.x parsing
[public->outside_call] continue=true
Dialplan: sofia/external/4435551212 at 140.239.xx.x Absolute Condition
[outside_call]
Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (PASS) [outside_call]
${module_exists(mod_cidlookup)}(true) =~ /true/ break=on-false
Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (PASS) [outside_call]
caller_id_name(4435551212) =~ /^4435551212$|^$/ break=on-false
Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (PASS) [outside_call]
caller_id_number(4435551212) =~ /^1?([2-9]\d\d[2-9]\d{6})$/ break=on-false
Dialplan: sofia/external/4435551212 at 140.239.xx.x Action
cidlookup(4435551212)
Dialplan: sofia/external/4435551212 at 140.239.xx.x parsing
[public->fix_cidnam_plus] continue=true
Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (FAIL)
[fix_cidnam_plus] caller_id_name(4435551212) =~
/^\+1?([2-9]\d\d[2-9]\d{6})$/ break=on-false
Dialplan: sofia/external/4435551212 at 140.239.xx.x parsing
[public->currently_running] continue=true
Dialplan: sofia/external/4435551212 at 140.239.xx.x Absolute Condition
[currently_running]
Dialplan: sofia/external/4435551212 at 140.239.xx.x Action info()
Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (FAIL)
[currently_running] caller_id_name(4435551212) =~ /^Currently running a
lookup/ break=on-false

Caller-Direction: [inbound]
Caller-Username: [4435551212]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [SMITH,JOHN]
Caller-Caller-ID-Number: [4435551212]
Caller-Network-Addr: [140.239.xx.x]
Caller-ANI: [4435551212]
Caller-Destination-Number: [4105551212]

_____________________________________________
From: Phil Quesinberry
Sent: Thursday, July 26, 2012 10:29 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: Setting effecting_caller_id_name


The reply is different each time, depending upon the number being looked up.
So, I just want to look at the first part of the string.  If FS can't do a
regex match without the trailing $, I'm guessing there's a way to just do it
in XML.

I'll try and see what I can find after the storm passes unless you have a
better idea, I need to shut this computer down right now.

Thanks,

- Phil


You can just not use a regex.

Do you need to escape the spaces?

Brian Foster
Endigo Computer LLC


_____________________________________________
From: Phil Quesinberry
Sent: Thursday, July 26, 2012 5:46 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: Setting effecting_caller_id_name


If you put the $ at the end then it will try to match the entire string
instead of just the beginning of it, which won't work in this case.  Is
there a way to match just the beginning of the string in FS?

Thanks,

- Phil


You need a $ after 'lookup' for it to be a regex.

Brian Foster
Endigo Computer LLC


_____________________________________________
From: Phil Quesinberry
Sent: Thursday, July 26, 2012 3:59 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Setting effecting_caller_id_name


And while I'm asking dumb questions.

When doing CNAM dips from opencnam.com, often you get a result of "Currently
running a lookup for phone 'xxxxxxxxxx'. on incoming calls, typically for
wireless or other unknown name callers and I wanted to change that to
"Wireless/Unknown"  Since caller_id_name is apparently read-only, I am
attempting to set effective_caller_id_name.  I put the following in
public.xml right below the "fix_cidnam_plus" entry, in other words after a
CNAM lookup has been performed.

   <extension name="fix_currently_running" continue="true">
      <condition field="caller_id_name" expression="^Currently running a
lookup">
        <action application="set"
data="effective_caller_id_name=Wireless/Unknown"/>
      </condition>
    </extension>

If I crafted my regex properly, then it should be matching on the first part
of the string and setting the variable appropriately.  Is
'effective_caller_id_name' the variable I should be setting?

Many thanks,

Phil Quesinberry
Q Systems Engineering, Inc.
Electronic Controls and Embedded Systems Development
(410) 969-8002
http://www.qsystemsengineering.com

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From ajohnston at blimessaging.com  Wed Aug  1 00:16:50 2012
From: ajohnston at blimessaging.com (Adam Johnston)
Date: Tue, 31 Jul 2012 16:16:50 -0400
Subject: [Freeswitch-users] FreeSWITCH dying with no core file
In-Reply-To: <5013012C.9050003@freeswitch.org>
References: <CAE2A9TCfrCw1=9ScVzsCcnP0CvaYVpMbHPAm19ChwY3rEQrN8A@mail.gmail.com>
	<5013012C.9050003@freeswitch.org>
Message-ID: <CAE2A9TDGczBNAss75AE5pVqMutotrkec7uvPWmJwjkV+t8+AZA@mail.gmail.com>

That did it. My sporadic crash happened again and, sure enough, there's a
core file.

Thanks again,
Adam Johnston

On Fri, Jul 27, 2012 at 4:59 PM, Stefan Knoblich <stkn at freeswitch.org>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> On 07/27/12 21:34, Adam Johnston wrote:
> > I use FreeSWITCH exclusively for high-volume fax, both sending and
> receiving, and I've been running into a sporadic issue where the FreeSWITCH
> process will die with nothing in the logs to
> > indicate an issue and no core file anywhere on the system. I'm setting
> "ulimit -c unlimited" in my init script, so I would expect to see a core
> file. Version: FreeSWITCH Version
> > 1.2.0-rc2+git~20120713T162602Z~0afd7318bd+unclean~20120713T182401Z
>
> You need to set /proc/sys/fs/suid_dumpable to a non-zero value (see man 5
> proc), to enable core dumping for processes
> that switch their UID on startup (such as freeswitch with -u option).
>
> By default, the freeswitch process will try to put the core file into the
> CWD (current work directory),
> so make sure the UID it is running as can write there (or "cd" into a
> different directory before starting it).
> You might also want to check out man 5 core, for all the other options
> (e.g. piping core dumps into a custom handler, with core_pattern).
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v2.0.19 (GNU/Linux)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
>
> iEYEARECAAYFAlATASwACgkQjiIIAK4rYUrjvQCgh1grtcBfPVhbZNw8tdQpsyda
> 614Amwb1pKDFL93dU33cy2wj1FNYYnzJ
> =0Xqk
> -----END PGP SIGNATURE-----
>
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From gabe at gundy.org  Wed Aug  1 03:31:18 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 31 Jul 2012 17:31:18 -0600
Subject: [Freeswitch-users] No call_uuid in ringing state unix odbc for
	PostgreSQL
In-Reply-To: <1343558012157-7581267.post@n2.nabble.com>
References: <1343558012157-7581267.post@n2.nabble.com>
Message-ID: <CAMwZ-t+3-6hSzzgYPivCE_bpboWWQwgnPCXg9nZqC6H2CY5spQ@mail.gmail.com>

On Sun, Jul 29, 2012 at 4:33 AM, cogs66 <cogs66 at gmail.com> wrote:
> I have tested on other FreeSWITCH installations and everything works great so i know it is
> to do with this rogue setup.

I just wanted to weigh in on this... It seems unlikely that ODBC and
PostgreSQL are at the root of this. Sorry, that's all I have for you
:) Let us know what you find out.

Best,
Gabe


From lloyd.aloysius at gmail.com  Wed Aug  1 09:47:44 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Wed, 1 Aug 2012 01:47:44 -0400
Subject: [Freeswitch-users] xml cdr help
In-Reply-To: <CC3D55FD.3ABE4%krice@freeswitch.org>
References: <CAGvuWYaGt0mCoNDA0o6NXnX2d94a98V+DupUgymNyGzUybjMxg@mail.gmail.com>
	<CC3D55FD.3ABE4%krice@freeswitch.org>
Message-ID: <CAGvuWYZ5nV=jmU-ktT23zrEe2dwAfmShmjr7Fo1BKYKL0mAsDA@mail.gmail.com>

Brian & Ken - Thank you for the detail explanation. I understand the
concept now.

1416471234  144  outbound   *7f055bf3-6622-4a47-8a86-e92c7683d74d*
898cc58c-8447-4a51-b559-3d1f2a063014
 2012/07/30 15:37:41 00:00:12

1416471234  144  inbound    898cc58c-8447-4a51-b559-3d1f2a063014 *
7f055bf3-6622-4a47-8a86-e92c7683d74d*  2012/07/301 5:37:23 00:00:30


See above the two records. I want to show only the following record in the
CDR report. Because that is time the user on the phone. Is there any
recommended way to filter the record from cdr table

1416471234  144  outbound   *7f055bf3-6622-4a47-8a86-e92c7683d74d*
 898cc58c-8447-4a51-b559-3d1f2a063014  2012/07/30 15:37:41 00:00:12

Thank you
Lloyd

On Tue, Jul 31, 2012 at 10:29 AM, Ken Rice <krice at freeswitch.org> wrote:

>  The Inbound/Outbound is from the perspective of FreeSWITCH... Keep in
> mind that FreeSWITCH is a b2bua that means that 1 call that goes Endpoint
> -> FreeSWITCH -> EndPoint is really 2 calls bridged together by freeswitch
> A Leg = Endpoint -> FreeSwitch (inbound on your CDR)  B Leg = FreeSWITCH ->
> Endpoint  (Outbound tagged on your CDR)
>
> You?ll notice that the UUIDs are flipped, cause each leg has a unique UUID
> and the ?Bridge UUID? field is the link back to the other leg(s) associated
> with that call
>
>
>
> On 7/31/12 8:37 AM, "Lloyd Aloysius" <lloyd.aloysius at gmail.com> wrote:
>
> Hi All:
>
> I implement the xml cdr. Now For a Incoming call -> IVR - > Dial Extension
> create two records see below. Basically both records should say inbound
> call . But one say as outbound. What is the reason the xml cdr module
> behave this way. Also for a one incoming call to a destination why showing
> the two records?
>
>
> Also I notice when we make a outbound call the direction field say
> inbound. But it should be outbound. Any help is appreciated.
>
> callerid number , destination number , uuid ,bridge uuid , date ,time
>
> 1416471234  144  outbound   *7f055bf3-6622-4a47-8a86-e92c7683d74d*898cc58c-8447-4a51-b559-3d1f2a063014  2012/07/30 15:37:41 00:00:12
>
> 1416471234  144  inbound    898cc58c-8447-4a51-b559-3d1f2a063014 *
> 7f055bf3-6622-4a47-8a86-e92c7683d74d*  2012/07/301 5:37:23 00:00:30
>
>
> Thanks
> Lloyd
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at gmail.com  Wed Aug  1 10:20:41 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Wed, 1 Aug 2012 02:20:41 -0400
Subject: [Freeswitch-users] ESL - phpmod compile Error
Message-ID: <CAGvuWYa7c7Fy+B8gKNU4cSFLT6HtjYpC_03UvqezRRJSw4dXhw@mail.gmail.com>

Hi All:

when I compile the ESL module for php . OS - Centos 6.2 / 64 bit

libs/esl

make phpmod

I got the following Errors.

 make phpmod
make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
CFLAGS="-I/usr/src/freeswitch.git/libs/esl/src/include -DHAVE_EDITLINE -g
-ggdb -I../../libs/libedit/src/ -fPIC -O2"
CXXFLAGS="-I/usr/src/freeswitch.git/libs/esl/src/include -DHAVE_EDITLINE -g
-ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php
make[1]: Entering directory `/usr/src/freeswitch.git/libs/esl/php'
g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib6464 -lcrypt
-lcrypt -ledit -lncurses -lm -ldl -lnsl -lm -ldl -ldl -lm -lcrypt -lm
-lcrypt -lpthread -o ESL.so -L.
/usr/bin/ld: cannot find -ledit
collect2: ld returned 1 exit status
make[1]: *** [ESL.so] Error 1
make[1]: Leaving directory `/usr/src/freeswitch.git/libs/esl/php'
make: *** [phpmod] Error 2


How to solve this issue.

Thanks
Lloyd
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From oseslija at gmail.com  Wed Aug  1 10:24:14 2012
From: oseslija at gmail.com (Ognjen Seslija)
Date: Wed, 1 Aug 2012 08:24:14 +0200
Subject: [Freeswitch-users] SIP to make phone reboot or resync
In-Reply-To: <CAKbxfG95Os62f7KRixUqsMdXbEbC8C1AznK9uUcV=oqOs2FV2Q@mail.gmail.com>
References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com>
	<CAKbxfG-KV=dTLQ=HP=ZsV_P1kp-NU0AyXczXbEciT4wLH9D2nw@mail.gmail.com>
	<094e01cd6c43$9e44cc90$dace65b0$@bizfocused.com>
	<CAKbxfG88tDW6ct8BpruBUX8Qh0uxPqOi8uowUU8RULKBvQrsKQ@mail.gmail.com>
	<0c4001cd6cdf$b2603a10$1720ae30$@bizfocused.com>
	<E7BDBB0E-E2D1-4F25-BF5C-8B8434203133@imtelecom.fr>
	<0deb01cd6d3f$67e43a30$37acae90$@bizfocused.com>
	<CAKbxfG95Os62f7KRixUqsMdXbEbC8C1AznK9uUcV=oqOs2FV2Q@mail.gmail.com>
Message-ID: <CAM21H-M2YYRax15gABr7Hs1WfoLDzTNPxh67+5AhbrO5TXLeoA@mail.gmail.com>

Didn't know this...so Tony you make prop things non-prop..that's like
Prometheus work :)

On Mon, Jul 30, 2012 at 11:43 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> reverse-auth-user and reverse-auth-pass params in the params section
> of the <user> xml tag in the user directory can be configured instead
> of a gateway and it will use those credentials instead.
>
>
> On Sat, Jul 28, 2012 at 11:05 PM, Sean Devoy <sdevoy at bizfocused.com>
> wrote:
> > Thank you sir.  That did help, I can resync my phone test either with
> > user/pass using any proxy user/pass from the lines on THAT phone.  I can
> > also disable Auth Resync Reboot and let anyone send reboots - which
> might be
> > fun, but seems like a bad plan!
> >
> > This is not very useful for me since the phone's proxy user/pass must be
> > specified in the gateway XML for whole realm in the the SIP profile.  I
> > hoped to be able to re-provision any phone on demand. It seems
> impractical
> > to have to restart/rescan my sofia gateway every time  I want to issue a
> > resync to a different phone.
> >
> > Am I missing something?
> >
> > Sean
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Vallimamod ABDULLAH
> > Sent: Saturday, July 28, 2012 1:05 PM
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync
> >
> > Hi,
> >
> > It's normally the sip account credentials. There is also an option in the
> > admin interface to disable this authentication ("Auth Resync-Reboot"
> option
> > if I recall correctly in Ext tab)
> >
> > Best Regards,
> > Vallimamod Abdullah
> > .
> >
> >
> > On Jul 28, 2012, at 6:40 PM, Sean Devoy wrote:
> >
> >> HI all,
> >>
> >> I am close, but I still don't understand what credentials are required.
> > The response is:
> >> SIP/2.0 401 Unauthorized
> >>
> >> I have tried the web admin credentials for the phone, I don't k now what
> > FS credentials I could pass.
> >>
> >> Any ideas?
> >>
> >> Thanks,
> >> Sean
> >>
> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> >> Anthony Minessale
> >> Sent: Friday, July 27, 2012 6:12 PM
> >> To: FreeSWITCH Users Help
> >> Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync
> >>
> >> Make a gateway with no reg and the credentials and name it after the
> realm
> > in the challenge.
> >>
> >> On Jul 27, 2012 5:05 PM, "Sean Devoy" <sdevoy at bizfocused.com> wrote:
> >> Thank Anthony, BUT.. Error:
> >> 2012-07-27 17:58:55.118642 [ERR] sofia_reg.c:2165 Cannot locate any
> >> authentication credentials to complete an authentication request for
> >> realm '"fs_bfis.bizfocused.com"'
> >>
> >> 1 - where do I specify the credentials?
> >> 2 - Are these Freeswitch credentials or phone credentials?  I suspect
> >> the latter.
> >>
> >> -----Original Message-----
> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> >> Anthony Minessale
> >> Sent: Friday, July 27, 2012 1:21 PM
> >> To: FreeSWITCH Users Help
> >> Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync
> >>
> >> sofia profile <prof name> check_sync [call_id]  // no call_id means
> >> all sofia profile <prof name> flush_inbound_reg [call_id] [reboot]
> >>
> >>
> >>
> >> On Fri, Jul 27, 2012 at 12:06 PM, Sean Devoy <sdevoy at bizfocused.com>
> > wrote:
> >> > HI All,
> >> >
> >> >
> >> >
> >> > Does anyone have an code that will cause Freeswitch to send a SIP
> >> > message to a CISCO SPA5xx phone that will cause it to reboot or
> >> > resync (aka re-provision)?  I know about the URLs that cause the
> >> > phone to do this, but they are NATed, so I need to use SIP to hit
> them.
> >> >
> >> >
> >> >
> >> > Assume I am programming language omni-lingual for this request!
> >> >
> >> >
> >> >
> >> > Thanks,
> >> >
> >> > Sean
> >> >
> >> >
> >> > ____________________________________________________________________
> >> > __ ___ Professional FreeSWITCH Consulting Services:
> >> > consulting at freeswitch.org
> >> > http://www.freeswitchsolutions.com
> >> >
> >> > 
> >> > 
> >> >
> >> > Official FreeSWITCH Sites
> >> > http://www.freeswitch.org
> >> > http://wiki.freeswitch.org
> >> > http://www.cluecon.com
> >> >
> >> > Join Us At ClueCon - Aug 7-9, 2012
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u
> >> > se
> >> > rs
> >> > http://www.freeswitch.org
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> ______________________________________________________________________
> >> ___ Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> rs
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >> ______________________________________________________________________
> >> ___ Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
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> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> rs
> >> http://www.freeswitch.org
> >> ______________________________________________________________________
> >> ___ Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> rs
> >> http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing 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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> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://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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>
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>
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From vbvbrj at gmail.com  Wed Aug  1 10:53:39 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Wed, 01 Aug 2012 09:53:39 +0300
Subject: [Freeswitch-users] $${base_dir} effect
Message-ID: <5018D273.1060308@gmail.com>

Hello.

I have a start up script which CDs to basedir of FS before launching 
daemon with "bin/freeswitch -nc".

In freeswitch.xml I have  <X-PRE-PROCESS cmd="set" data="base_dir=."/> 
to set other paths relative to current directory. As in default config 
files from repository most parameters which uses $${base_dir}/some-path 
a commented out, its hard to test. I've encountered problem with 
mod_callcenter. In callcenter.conf.xml there is:
<param name="record-template" 
value="$${base_dir}/${destination_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.${caller_id_number}.${uuid}.wav"/>. 
In final freeswitch.xml.fsxml the line is:
<param name="record-template" 
value="/srv/freeswitch/recordings/${destination_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.${caller_id_number}.${uuid}.wav"/>
But final recorded path from logs become 
$${sounds_dir}/$${base_dir}/recordings/ ie 
/opt/freeswitch/sounds/en/us/callie/./recordings/ or 
/opt/freeswitch/sounds/en/us/callie/recordings/ in final creation.

Where does this $${sounds_dir} come if I don't specify it?

Thank you.


From ramesh_mind at yahoo.com  Wed Aug  1 11:11:16 2012
From: ramesh_mind at yahoo.com (ramesh)
Date: Wed, 1 Aug 2012 00:11:16 -0700 (PDT)
Subject: [Freeswitch-users] Call is Automatically Retried for 3 times,
	if call is unanswered
Message-ID: <1343805076936-7581401.post@n2.nabble.com>

Hi Team, 

I made a script to dial a user , and if not answered  the second user should
get the dial. Everything works file , Except the first user is dialed for 3
times , if the user is busy or unanswered , i was trying to figure out what
could be the problem for this issue, but i couldn't .

Can Anyone Figure out what is the reason for this issue.
below is my lua dial script 

session:execute("bridge","{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true}sofia/gateway/bandwidth.com/+919xxxxxxxx|sofia/gateway/bandwidth.com/+919xxxxxxxx");


following is the log i can get.

Any Help or suggestion would be really helpful!

[NOTICE] sofia.c:5594 Ring-Ready sofia/internal/+919677080275!
2012-08-01 07:09:13.212635 [DEBUG] switch_ivr_originate.c:1164 Raw Codec
Activation Success L16 at 16000hz 1 channel 20ms
2012-08-01 07:09:13.212635 [DEBUG] switch_core_codec.c:216
rtmp/default/+542914850488 Push codec L16:70
2012-08-01 07:09:13.212635 [DEBUG] switch_ivr_originate.c:1227 Play Ringback
Tone [%(2000,4000,440,480)]
2012-08-01 07:09:16.532655 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK]
2012-08-01 07:09:16.532655 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK]
2012-08-01 07:09:16.552637 [DEBUG] sofia.c:5502 Channel
sofia/internal/+919677080275 entering state [proceeding][183]
2012-08-01 07:09:16.552637 [DEBUG] sofia.c:5513 Remote SDP:
v=0
o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14
s=SIP Media Capabilities
c=IN IP4 65.115.130.14
t=0 0
m=audio 10382 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[SPEEX:99:16000:20:42200]
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000]
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000]
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:2919 Set Codec
sofia/internal/+919677080275 PCMU/8000 20 ms 160 samples 64000 bits
2012-08-01 07:09:16.552637 [DEBUG] switch_core_codec.c:111
sofia/internal/+919677080275 Original read codec set to PCMU:0
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
payload to 101
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3171 AUDIO RTP
[sofia/internal/+919677080275] 10.244.15.45 port 30506 -> 65.115.130.14 port
10382 codec: 0 ms: 20
2012-08-01 07:09:16.552637 [DEBUG] switch_rtp.c:1661 Starting timer [soft]
160 bytes per 20ms
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send
payload to 101
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive
payload to 101
2012-08-01 07:09:16.552637 [NOTICE] sofia_glue.c:3945 Pre-Answer
sofia/internal/+919677080275!
2012-08-01 07:09:16.552637 [DEBUG] switch_channel.c:2936
(sofia/internal/+919677080275) Callstate Change RINGING -> EARLY
2012-08-01 07:09:16.552637 [DEBUG] switch_channel.c:2978 Send signal
rtmp/default/+542914850488 [BREAK]
2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2782 Adding tone spec
480,620 index 0 hits 1
2012-08-01 07:09:16.572648 [DEBUG] switch_core_media_bug.c:457 Attaching BUG
to sofia/internal/+919677080275
2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2782 Adding tone spec
1776.7 index 1 hits 2
2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2877
sofia/internal/+919677080275 bug already running
2012-08-01 07:09:16.572648 [DEBUG] switch_rtp.c:3205 Correct ip/port
confirmed.
2012-08-01 07:09:16.572648 [DEBUG] switch_core_io.c:340 Setting BUG Codec
PCMU:0
2012-08-01 07:09:33.072650 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK]
2012-08-01 07:09:33.072650 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK]
2012-08-01 07:09:33.092641 [DEBUG] sofia.c:5502 Channel
sofia/internal/+919677080275 entering state [proceeding][183]
2012-08-01 07:09:33.092641 [DEBUG] sofia.c:5510 Duplicate SDP
v=0
o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14
s=SIP Media Capabilities
c=IN IP4 65.115.130.14
t=0 0
m=audio 10382 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:2853 Already using PCMU
2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
payload to 101
2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:3141 Audio params are
unchanged for sofia/internal/+919677080275.
2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:3151
sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0
2012-08-01 07:09:36.372634 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK]
2012-08-01 07:09:36.372634 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK]
2012-08-01 07:09:36.392686 [DEBUG] sofia.c:5502 Channel
sofia/internal/+919677080275 entering state [proceeding][183]
2012-08-01 07:09:36.392686 [DEBUG] sofia.c:5510 Duplicate SDP
v=0
o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14
s=SIP Media Capabilities
c=IN IP4 65.115.130.14
t=0 0
m=audio 10382 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:2853 Already using PCMU
2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
payload to 101
2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:3141 Audio params are
unchanged for sofia/internal/+919677080275.
2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:3151
sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0
2012-08-01 07:09:38.512638 [DEBUG] libdingaling.c:1624 Sent keep alive
signal
2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK]
2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK]
2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK]
2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK]
2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5502 Channel
sofia/internal/+919677080275 entering state [proceeding][183]
2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5510 Duplicate SDP
v=0
o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14
s=SIP Media Capabilities
c=IN IP4 65.115.130.14
t=0 0
m=audio 10382 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:2853 Already using PCMU
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
payload to 101
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3141 Audio params are
unchanged for sofia/internal/+919677080275.
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3151
sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0
2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5502 Channel
sofia/internal/+919677080275 entering state [proceeding][183]
2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5510 Duplicate SDP
v=0
o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14
s=SIP Media Capabilities
c=IN IP4 65.115.130.14
t=0 0
m=audio 10382 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:2853 Already using PCMU
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
payload to 101
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3141 Audio params are
unchanged for sofia/internal/+919677080275.
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3151
sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0
2012-08-01 07:10:11.012635 [DEBUG] switch_core_codec.c:241
rtmp/default/+542914850488 Restore previous codec SPEEX:99.
2012-08-01 07:10:11.012635 [DEBUG] switch_channel.c:2852
(sofia/internal/+919677080275) Callstate Change EARLY -> HANGUP
2012-08-01 07:10:11.012635 [NOTICE] switch_ivr_originate.c:3182 Hangup
sofia/internal/+919677080275 [CS_CONSUME_MEDIA] [NO_ANSWER]


Thanks
Ramesh







--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Call-is-Automatically-Retried-for-3-times-if-call-is-unanswered-tp7581401.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From peter.olsson at visionutveckling.se  Wed Aug  1 11:18:12 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 1 Aug 2012 07:18:12 +0000
Subject: [Freeswitch-users] ESL - phpmod compile Error
Message-ID: <1FFF97C269757C458224B7C895F35F1513B61A@cantor.std.visionutv.se>

Submit to http://jira.freeswitch.org

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Lloyd Aloysius
Skickat: den 1 augusti 2012 08:21
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] ESL - phpmod compile Error

Hi All:

when I compile the ESL module for php . OS - Centos 6.2 / 64 bit

libs/esl

make phpmod

I got the following Errors.

 make phpmod
make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch.git/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2" CXXFLAGS="-I/usr/src/freeswitch.git/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php
make[1]: Entering directory `/usr/src/freeswitch.git/libs/esl/php'
g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib6464 -lcrypt -lcrypt -ledit -lncurses -lm -ldl -lnsl -lm -ldl -ldl -lm -lcrypt -lm -lcrypt -lpthread -o ESL.so -L.
/usr/bin/ld: cannot find -ledit
collect2: ld returned 1 exit status
make[1]: *** [ESL.so] Error 1
make[1]: Leaving directory `/usr/src/freeswitch.git/libs/esl/php'
make: *** [phpmod] Error 2


How to solve this issue.

Thanks
Lloyd
!DSPAM:5018c8de32761380847499!
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From slava at tangramltd.com  Wed Aug  1 12:20:57 2012
From: slava at tangramltd.com (Viacheslav Dubrovskyi)
Date: Wed, 01 Aug 2012 11:20:57 +0300
Subject: [Freeswitch-users] $${base_dir} effect
In-Reply-To: <5018D273.1060308@gmail.com>
References: <5018D273.1060308@gmail.com>
Message-ID: <5018E6E9.2000309@tangramltd.com>

01.08.2012 09:53, Vbvbrj ?????:
> Where does this $${sounds_dir} come if I don't specify it?
in configure.in

AC_ARG_WITH([soundsdir],
[AS_HELP_STRING([--with-soundsdir=DIR], [Put sound files into this
location (default: $prefix/sounds)])], [soundsdir="$withval"],
[soundsdir="$prefix/sounds"])
AC_SUBST(soundsdir)


So you need setup it during build: -

./configure --with-soundsdir ...


-- 
WBR,
Viacheslav Dubrovskyi


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From vbvbrj at gmail.com  Wed Aug  1 12:29:10 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Wed, 01 Aug 2012 11:29:10 +0300
Subject: [Freeswitch-users] $${base_dir} effect
In-Reply-To: <5018E6E9.2000309@tangramltd.com>
References: <5018D273.1060308@gmail.com> <5018E6E9.2000309@tangramltd.com>
Message-ID: <5018E8D6.4060906@gmail.com>

On 01.08.2012 11:20, Viacheslav Dubrovskyi wrote:
> 01.08.2012 09:53, Vbvbrj ?????:
>> Where does this $${sounds_dir} come if I don't specify it?
> in configure.in
>
> AC_ARG_WITH([soundsdir],
> [AS_HELP_STRING([--with-soundsdir=DIR], [Put sound files into this
> location (default: $prefix/sounds)])], [soundsdir="$withval"],
> [soundsdir="$prefix/sounds"])
> AC_SUBST(soundsdir)
>
>
> So you need setup it during build: -
>
> ./configure --with-soundsdir ...

On configure, default is:
soundsdir = ${prefix}/sounds
recordingsdir = ${prefix}/recordings

Also I specify soundsdir and recordingsdir using command line on FS start.

If in callcenter.conf.xml I use absolute path for recordings folder, the 
resulting folder is correct. If I use relative, then value of 
${sounds_dir} is prepended. It should not do this. Inot prepend 
${base_dir} ?


From X.Liu at hw.ac.uk  Wed Aug  1 12:48:46 2012
From: X.Liu at hw.ac.uk (Liu, Xingkun)
Date: Wed, 1 Aug 2012 09:48:46 +0100
Subject: [Freeswitch-users] Am I using play_and_detect_speech correctly?
References: <D346F0E06A1D9341B7FF1FC1B525698C5C128A@ex10.mail.win.hw.ac.uk><CAD-qRkhF-=6y6xWK0eyuZN3FSCUw7qAhHwyHbnJnXTmoZxZXrA@mail.gmail.com><50179069.6020700@hw.ac.uk>
	<CAD-qRkiL-KFF-tkKk3Ni=eOn+A6dp5eKxe61PK=6_m5N_WBaFw@mail.gmail.com>
Message-ID: <D346F0E06A1D9341B7FF1FC1B525698C5C1291@ex10.mail.win.hw.ac.uk>

Hi Chris,

Many thanks for your info and suggestions!

Would you please tell me a bit more or any clue about the idea of using detect_speech APP to handle
barge-in over ESL? It is an interesting idea to try but for now I don't have any clue how it could be
done over Java ESL without touching or modifying detect_speech APP codes.

Thanks again!

Best regards,

Xing




-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Christopher Rienzo
Sent: Tue 7/31/2012 13:44
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Am I using play_and_detect_speech correctly?
 
I looked over the code again last night and verified that
play_and_detect_speech should try to stop detection before loading the
grammar.  However, that scenario is not really tested, so try the
workaround first.  If that doesn't work, open a jira ticket with detailed
logs so it can be fixed.

You can use the detect_speech APP to do everything over ESL.  It is a lot
more work, but once you figure it out, you have the freedom to do whatever
you want, including handling barge-in.  play_and_detect_speech is designed
to handle the typical use case.

Chris



On Tue, Jul 31, 2012 at 3:59 AM, x.liu <x.liu at hw.ac.uk> wrote:

>  Hi Chris,
>
> Okay, thanks! I will have tries to see how it works.
>
> For the normal use of the app: play_and_detect_speech, will it resume
> previous ASR session,
> or will it stop previous session and restart a new session each time when
> I issue it via ESL?
>
> For the app of detect_speech, I can first to start an ASR session by
> "detect_speech speechMod grammar grammarPath",
> then I can do "detect_speech pause", or "detect_speech resume" or
> "detect_speech stop".
> But for play_and_detect_speech, I am not sure how it works in terms of
> starting, pause, resume/restart sequence.
>
> The play_and_detect_speech is a very good, useful app as it supports the
> barge-in. So we definitely need to use it.
>
> Thanks!
>
> Xing
>
>
>
>
>
> On 07/31/2012 12:53 AM, Christopher Rienzo wrote:
>
> You are using that APP in a way that hasn't been tested.  Try to do
> "speech_detect pause" and see if that helps.  It will reserve your ASR
> session for reuse while "speech_detect stop" will tear it down completely.
>  If that doesn't work, open a jira ticket and attach a full call trace.
>  It's difficult to understand exactly what is happening from your
> description.
>
>
>  Chris
>
>
>
> On Mon, Jul 30, 2012 at 5:38 PM, Liu, Xingkun <X.Liu at hw.ac.uk> wrote:
>
>>  Hello,
>>
>> I am using play_and_detect_speech with Java ESL in my IVR applications.
>>
>> Previously I call it again each time after I receive any recognition
>> event,
>> like recognition complete, no-input-timeout, or recognition-timeout,
>> it seems to work fine.
>>
>> Now I have changed my app to issue play_and_detect_speech command based on
>> my available system utterances as well as the speech event.
>>
>> I.e., I use a separate thread to constantly check if there is a system
>> utterance coming in
>> from another component of my application, if there is any utterance I
>> issue the command which will
>> speak the new utterance and listen to user input no matter whether or not
>> previous
>> command has finished. And if there is any speech event (recognition
>> result, timeout etc.)
>> the play_and_detect_speech command is also issued but with playing
>> silence.
>>
>> Obviously the new command will stop the utterance speaking of the
>> previous command if it is not finished.
>>
>> My question is
>>
>> will the new play_and_detect_speech command also stop the previous ASR
>> listening
>> or will there be many ASR listening channel and sending speech data (or
>> silence) to ASR server?
>>
>> Do I need to explicitly issue a "stop" commnad before issuing a new
>> play_and_detect_speech?
>> If yes, how to do that, by "detect_speech stop"?
>>
>> Recently there is a network traffice problem (lots of connections /data
>> transportation to the ASR server machine)
>> when I am running my application. I am not sure if this is because of
>> other issues
>> or because of my new changes to the way of using play_and_detect_speech.
>>
>> Please any one could shed a light on this?
>>
>> Many thanks!
>>
>> Xing
>>
>>
>> ------------------------------
>>
>>  *Heriot-Watt University is the Sunday Times Scottish University of the
>> Year 2011-2012.*
>>
>>  We invite research leaders and ambitious early career researchers to
>> join us in leading and driving research in key inter-disciplinary themes.
>> Please see www.hw.ac.uk/researchleaders for further information and how
>> to apply.
>>
>>  Heriot-Watt University is a Scottish charity registered under charity
>> number SC000278.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
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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
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>
>
> ------------------------------
>
>  *Heriot-Watt University is the Sunday Times Scottish University of the
> Year 2011-2012.*
>
>  We invite research leaders and ambitious early career researchers to join
> us in leading and driving research in key inter-disciplinary themes. Please
> see www.hw.ac.uk/researchleaders for further information and how to
> apply.
>
>  Heriot-Watt University is a Scottish charity registered under charity
> number SC000278.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing 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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>
>



-- 
Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012

We invite research leaders and ambitious early career researchers to 
join us in leading and driving research in key inter-disciplinary themes. 
Please see www.hw.ac.uk/researchleaders for further information and how
to apply.

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From ramesh_mind at yahoo.com  Wed Aug  1 13:15:53 2012
From: ramesh_mind at yahoo.com (ramesh)
Date: Wed, 1 Aug 2012 02:15:53 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch Auto Retrying a call for 3 times ,
	if it is not answered
Message-ID: <1343812553346-7581406.post@n2.nabble.com>

Hi Team, 

I made a script to dial a user , and if not answered  the second user should
get the dial. Everything works file , Except the first user is dialed for 3
times , if the user is busy or unanswered , i was trying to figure out what
could be the problem for this issue, but i couldn't . 

Can Anyone Figure out what is the reason for this issue. 
below is my lua dial script 

session:execute("bridge","{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true}sofia/gateway/bandwidth.com/+919xxxxxxxx|sofia/gateway/bandwidth.com/+919xxxxxxxx");


following is the log i can get. 

Any Help or suggestion would be really helpful! 

[NOTICE] sofia.c:5594 Ring-Ready sofia/internal/+919677080275! 
2012-08-01 07:09:13.212635 [DEBUG] switch_ivr_originate.c:1164 Raw Codec
Activation Success L16 at 16000hz 1 channel 20ms 
2012-08-01 07:09:13.212635 [DEBUG] switch_core_codec.c:216
rtmp/default/+542914850488 Push codec L16:70 
2012-08-01 07:09:13.212635 [DEBUG] switch_ivr_originate.c:1227 Play Ringback
Tone [%(2000,4000,440,480)] 
2012-08-01 07:09:16.532655 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK] 
2012-08-01 07:09:16.532655 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK] 
2012-08-01 07:09:16.552637 [DEBUG] sofia.c:5502 Channel
sofia/internal/+919677080275 entering state [proceeding][183] 
2012-08-01 07:09:16.552637 [DEBUG] sofia.c:5513 Remote SDP: 
v=0 
o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 
s=SIP Media Capabilities 
c=IN IP4 65.115.130.14 
t=0 0 
m=audio 10382 RTP/AVP 0 101 
a=rtpmap:0 PCMU/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-15 
a=ptime:20 

2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[SPEEX:99:16000:20:42200] 
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:2919 Set Codec
sofia/internal/+919677080275 PCMU/8000 20 ms 160 samples 64000 bits 
2012-08-01 07:09:16.552637 [DEBUG] switch_core_codec.c:111
sofia/internal/+919677080275 Original read codec set to PCMU:0 
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
payload to 101 
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3171 AUDIO RTP
[sofia/internal/+919677080275] 10.244.15.45 port 30506 -> 65.115.130.14 port
10382 codec: 0 ms: 20 
2012-08-01 07:09:16.552637 [DEBUG] switch_rtp.c:1661 Starting timer [soft]
160 bytes per 20ms 
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send
payload to 101 
2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive
payload to 101 
2012-08-01 07:09:16.552637 [NOTICE] sofia_glue.c:3945 Pre-Answer
sofia/internal/+919677080275! 
2012-08-01 07:09:16.552637 [DEBUG] switch_channel.c:2936
(sofia/internal/+919677080275) Callstate Change RINGING -> EARLY 
2012-08-01 07:09:16.552637 [DEBUG] switch_channel.c:2978 Send signal
rtmp/default/+542914850488 [BREAK] 
2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2782 Adding tone spec
480,620 index 0 hits 1 
2012-08-01 07:09:16.572648 [DEBUG] switch_core_media_bug.c:457 Attaching BUG
to sofia/internal/+919677080275 
2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2782 Adding tone spec
1776.7 index 1 hits 2 
2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2877
sofia/internal/+919677080275 bug already running 
2012-08-01 07:09:16.572648 [DEBUG] switch_rtp.c:3205 Correct ip/port
confirmed. 
2012-08-01 07:09:16.572648 [DEBUG] switch_core_io.c:340 Setting BUG Codec
PCMU:0 
2012-08-01 07:09:33.072650 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK] 
2012-08-01 07:09:33.072650 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK] 
2012-08-01 07:09:33.092641 [DEBUG] sofia.c:5502 Channel
sofia/internal/+919677080275 entering state [proceeding][183] 
2012-08-01 07:09:33.092641 [DEBUG] sofia.c:5510 Duplicate SDP 
v=0 
o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 
s=SIP Media Capabilities 
c=IN IP4 65.115.130.14 
t=0 0 
m=audio 10382 RTP/AVP 0 101 
a=rtpmap:0 PCMU/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-15 
a=ptime:20 

2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 
2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:2853 Already using PCMU 
2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
payload to 101 
2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:3141 Audio params are
unchanged for sofia/internal/+919677080275. 
2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:3151
sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0 
2012-08-01 07:09:36.372634 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK] 
2012-08-01 07:09:36.372634 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK] 
2012-08-01 07:09:36.392686 [DEBUG] sofia.c:5502 Channel
sofia/internal/+919677080275 entering state [proceeding][183] 
2012-08-01 07:09:36.392686 [DEBUG] sofia.c:5510 Duplicate SDP 
v=0 
o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 
s=SIP Media Capabilities 
c=IN IP4 65.115.130.14 
t=0 0 
m=audio 10382 RTP/AVP 0 101 
a=rtpmap:0 PCMU/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-15 
a=ptime:20 

2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 
2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:2853 Already using PCMU 
2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
payload to 101 
2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:3141 Audio params are
unchanged for sofia/internal/+919677080275. 
2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:3151
sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0 
2012-08-01 07:09:38.512638 [DEBUG] libdingaling.c:1624 Sent keep alive
signal 
2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK] 
2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK] 
2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK] 
2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal
sofia/internal/+919677080275 [BREAK] 
2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5502 Channel
sofia/internal/+919677080275 entering state [proceeding][183] 
2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5510 Duplicate SDP 
v=0 
o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 
s=SIP Media Capabilities 
c=IN IP4 65.115.130.14 
t=0 0 
m=audio 10382 RTP/AVP 0 101 
a=rtpmap:0 PCMU/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-15 
a=ptime:20 

2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:2853 Already using PCMU 
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
payload to 101 
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3141 Audio params are
unchanged for sofia/internal/+919677080275. 
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3151
sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0 
2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5502 Channel
sofia/internal/+919677080275 entering state [proceeding][183] 
2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5510 Duplicate SDP 
v=0 
o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 
s=SIP Media Capabilities 
c=IN IP4 65.115.130.14 
t=0 0 
m=audio 10382 RTP/AVP 0 101 
a=rtpmap:0 PCMU/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-15 
a=ptime:20 

2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:2853 Already using PCMU 
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
payload to 101 
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3141 Audio params are
unchanged for sofia/internal/+919677080275. 
2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3151
sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0 
2012-08-01 07:10:11.012635 [DEBUG] switch_core_codec.c:241
rtmp/default/+542914850488 Restore previous codec SPEEX:99. 
2012-08-01 07:10:11.012635 [DEBUG] switch_channel.c:2852
(sofia/internal/+919677080275) Callstate Change EARLY -> HANGUP 
2012-08-01 07:10:11.012635 [NOTICE] switch_ivr_originate.c:3182 Hangup
sofia/internal/+919677080275 [CS_CONSUME_MEDIA] [NO_ANSWER] 


Thanks 
Ramesh 




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Auto-Retrying-a-call-for-3-times-if-it-is-not-answered-tp7581406.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ale975 at gmail.com  Wed Aug  1 15:40:38 2012
From: ale975 at gmail.com (Ale)
Date: Wed, 1 Aug 2012 13:40:38 +0200
Subject: [Freeswitch-users] freeswitch_licence_server problem
Message-ID: <CAB+ia=2DkqDXvPZEvRTWqVYF=1i3S4ViryEbwiJM9FxrtM9=Wg@mail.gmail.com>

Hello,

short: after installation fs show "Can't contact licence server." and
freeswitch_licence_server in console show "Unrecognised resource
G.729A/0".

I've purchased a license, downloaded 194 installer, mod_com_g729.so is
in lib64/freeswitch/mod dir, and license file without licenze.zip is
in /etc/freeswitch.
Freeswitch run as user freeswitch user on a centos.
Where i unload mod_g729 and load the mod_com, server correctly starts.
I also try to kill the server,
and manually start it a root user, but nothing change.

could someone give me any hints?
Thanks alessandro


From nathandownes at hotmail.com  Wed Aug  1 17:04:17 2012
From: nathandownes at hotmail.com (Mr Nathan Downes)
Date: Wed, 1 Aug 2012 23:04:17 +1000
Subject: [Freeswitch-users] FW: multi-tenant setup - call goes to wrong place
In-Reply-To: <03e401cd6be9$52f19b10$f8d4d130$@hotmail.com>
References: <03e401cd6be9$52f19b10$f8d4d130$@hotmail.com>
Message-ID: <BLU170-DS2348B20FE401112A311FEAD7C40@phx.gbl>

Hi,

 

I ended up locating the problem.. it appears when I was playing with
shared_presence.. to try and get BLF working for certain tenants. What I
thought would help from googling caused this issue somehow, below is what I
enabled to cause the problem.

 

In internal.xml

 

<!--enable to use presence -->

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

    <!-- send a presence probe on each register to query devices to send
presence instead of sending presence with less info -->

    <!--<param name="presence-probe-on-register" value="true"/>-->
this one enabled

    <!--<param name="manage-shared-appearance" value="false"/>-->
this one true + enabled

    <!-- used to share presence info across sofia profiles -->

    <!-- Name of the db to use for this profile -->

    <!-- param name="dbname" value="shared_presence"/> -->
this one  + enabled

    <param name="presence-hosts"
value="domain1.com,domain2.com,domain3.com,$${local_ip_v4}"/>

 

Is this actually required to use BLF for a certain domain?

 

 

From: Mr Nathan Downes [mailto:nathandownes at hotmail.com] 
Sent: Friday, 27 July 2012 9:17 PM
To: 'FreeSWITCH Users Help'
Subject: multi-tenant setup - call goes to wrong place

 

Hi,

 

I have a multi tenant setup with fusionpbx, for one domain I use extensions
that don't register just for voicemail i.e 201 at domain1.com , when someone
from xx at domain1.com calls 201 everything appears to go correctly, domain is
set and call goes to 201 at domain1.com but when it first becomes an ip it
sends to 201 at domain2.ip.address.  It doesn't seem to happen with all
extensions numbered the same in different domains..

 

Any idea what to look for??

 

Log as follows.

 

2012-07-26 09:37:35.178096 [NOTICE] switch_channel.c:926 New Channel
sofia/internal/306 at domain1.com [b62db898-d6b1-11e1-be15-73f389528c7f]

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/306 at domain1.com) Running State Change CS_NEW

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:403
(sofia/internal/306 at domain1.com) State NEW

2012-07-26 09:37:35.178096 [DEBUG] sofia.c:5838 Channel
sofia/internal/306 at domain1.com entering state [received][100]

2012-07-26 09:37:35.178096 [DEBUG] sofia.c:5849 Remote SDP:

v=0

o=- 60255882 60255882 IN IP4 115.64.93.203

s=-

c=IN IP4 115.64.93.203

t=0 0

m=audio 55938 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

 

2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash

2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3919 Deciding whether to
pass zrtp-hash between legs

2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ
not set, so not propagating zrtp-hash

2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:5034 Audio Codec Compare
[PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]

2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3020 Set Codec
sofia/internal/306 at domain1.com PCMA/8000 20 ms 160 samples 64000 bits

2012-07-26 09:37:35.178096 [DEBUG] switch_core_codec.c:111
sofia/internal/306 at domain1.com Original read codec set to PCMA:8

2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:5155 Set 2833 dtmf send/recv
payload to 101

2012-07-26 09:37:35.178096 [DEBUG] sofia.c:6077
(sofia/internal/306 at domain1.com) State Change CS_NEW -> CS_INIT

2012-07-26 09:37:35.178096 [DEBUG] switch_core_session.c:1228 Send signal
sofia/internal/306 at domain1.com [BREAK]

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/306 at domain1.com) Running State Change CS_INIT

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/306 at domain1.com) State INIT

2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:85
sofia/internal/306 at domain1.com SOFIA INIT

2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:125
(sofia/internal/306 at domain1.com) State Change CS_INIT -> CS_ROUTING

2012-07-26 09:37:35.178096 [DEBUG] switch_core_session.c:1228 Send signal
sofia/internal/306 at domain1.com [BREAK]

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/306 at domain1.com) State INIT going to sleep

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/306 at domain1.com) Running State Change CS_ROUTING

2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1919
(sofia/internal/306 at domain1.com) Callstate Change DOWN -> RINGING

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:433
(sofia/internal/306 at domain1.com) State ROUTING

2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:148
sofia/internal/306 at domain1.com SOFIA ROUTING

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:104
sofia/internal/306 at domain1.com Standard ROUTING

2012-07-26 09:37:35.178096 [INFO] mod_dialplan_xml.c:485 Processing
0286249343 <306>->201 in context domain1.com

Dialplan: sofia/internal/306 at domain1.com parsing [domain1.com->unloop]
continue=false

Dialplan: sofia/internal/306 at domain1.com parsing
[domain1.com->Local_Extension] continue=false

Dialplan: sofia/internal/306 at domain1.com Regex (PASS) [Local_Extension]
destination_number(201) =~ /(^\d{2,7}$)/ break=on-false

Dialplan: sofia/internal/306 at domain1.com Action set(dialed_extension=201) 

Dialplan: sofia/internal/306 at domain1.com Action export(dialed_extension=201)


Dialplan: sofia/internal/306 at domain1.com Action limit(hash ${domain_name}
201 ${limit_max} ${limit_destination}) 

Dialplan: sofia/internal/306 at domain1.com Action bind_meta_app(1 b s
execute_extension::dx XML features) 

Dialplan: sofia/internal/306 at domain1.com Action bind_meta_app(2 b s
record_session::/usr/local/freeswitch/recordings/archive/${strftime(%Y)}/${s
trftime(%b)}/${strftime(%d)}/${uuid}.wav) 

Dialplan: sofia/internal/306 at domain1.com Action bind_meta_app(3 b s
execute_extension::cf XML features) 

Dialplan: sofia/internal/306 at domain1.com Action bind_meta_app(4 b s
execute_extension::att_xfer XML features) 

Dialplan: sofia/internal/306 at domain1.com Action set(ringback=${us-ring}) 

Dialplan: sofia/internal/306 at domain1.com Action
set(transfer_ringback=local_stream://moh) 

Dialplan: sofia/internal/306 at domain1.com Action set(call_timeout=30) 

Dialplan: sofia/internal/306 at domain1.com Action
set(hangup_after_bridge=true) 

Dialplan: sofia/internal/306 at domain1.com Action set(continue_on_fail=true) 

Dialplan: sofia/internal/306 at domain1.com Action
hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe
r}) 

Dialplan: sofia/internal/306 at domain1.com Action
hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) 

Dialplan: sofia/internal/306 at domain1.com Action
set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}) 

Dialplan: sofia/internal/306 at domain1.com Action
hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) 

Dialplan: sofia/internal/306 at domain1.com Action
bridge(user/${user_data(${destination_number}@${domain_name} attr
id)}@${domain_name}) 

Dialplan: sofia/internal/306 at domain1.com Action answer() 

Dialplan: sofia/internal/306 at domain1.com Action sleep(1000) 

Dialplan: sofia/internal/306 at domain1.com Action voicemail(default
${domain_name} ${dialed_extension}) 

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:154
(sofia/internal/306 at domain1.com) State Change CS_ROUTING -> CS_EXECUTE

2012-07-26 09:37:35.178096 [DEBUG] switch_core_session.c:1228 Send signal
sofia/internal/306 at domain1.com [BREAK]

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:433
(sofia/internal/306 at domain1.com) State ROUTING going to sleep

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/306 at domain1.com) Running State Change CS_EXECUTE

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:440
(sofia/internal/306 at domain1.com) State EXECUTE

2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:241
sofia/internal/306 at domain1.com SOFIA EXECUTE

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:196
sofia/internal/306 at domain1.com Standard EXECUTE

EXECUTE sofia/internal/306 at domain1.com set(call_direction=local)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [call_direction]=[local]

EXECUTE sofia/internal/306 at domain1.com set(open=true)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [open]=[true]

EXECUTE sofia/internal/306 at domain1.com
hash(insert/domain1.com-spymap/306/b62db898-d6b1-11e1-be15-73f389528c7f)

EXECUTE sofia/internal/306 at domain1.com
hash(insert/domain1.com-last_dial/306/201)

EXECUTE sofia/internal/306 at domain1.com
hash(insert/domain1.com-last_dial/global/b62db898-d6b1-11e1-be15-73f389528c7
f)

EXECUTE sofia/internal/306 at domain1.com set(RFC2822_DATE=Thu, 26 Jul 2012
09:37:35 +1000)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [RFC2822_DATE]=[Thu, 26 Jul 2012 09:37:35
+1000]

EXECUTE sofia/internal/306 at domain1.com set(dialed_extension=201)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [dialed_extension]=[201]

EXECUTE sofia/internal/306 at domain1.com export(dialed_extension=201)

2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1093 EXPORT
(export_vars) [dialed_extension]=[201]

EXECUTE sofia/internal/306 at domain1.com limit(hash domain1.com 201 2 !BUSY)

2012-07-26 09:37:35.178096 [INFO] switch_limit.c:126 incr called:
domain1.com_201 max:2, interval:0

2012-07-26 09:37:35.178096 [INFO] mod_hash.c:202 Usage for domain1.com_201
is now 1/2

EXECUTE sofia/internal/306 at domain1.com bind_meta_app(1 b s
execute_extension::dx XML features)

2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *1
execute_extension::dx XML features

EXECUTE sofia/internal/306 at domain1.com bind_meta_app(2 b s
record_session::/usr/local/freeswitch/recordings/archive/2012/Jul/26/b62db89
8-d6b1-11e1-be15-73f389528c7f.wav)

2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *2
record_session::/usr/local/freeswitch/recordings/archive/2012/Jul/26/b62db89
8-d6b1-11e1-be15-73f389528c7f.wav

EXECUTE sofia/internal/306 at domain1.com bind_meta_app(3 b s
execute_extension::cf XML features)

2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *3
execute_extension::cf XML features

EXECUTE sofia/internal/306 at domain1.com bind_meta_app(4 b s
execute_extension::att_xfer XML features)

2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *4
execute_extension::att_xfer XML features

EXECUTE sofia/internal/306 at domain1.com set(ringback=%(2000, 4000, 440.0,
480.0))

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [ringback]=[%(2000, 4000, 440.0, 480.0)]

EXECUTE sofia/internal/306 at domain1.com
set(transfer_ringback=local_stream://moh)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [transfer_ringback]=[local_stream://moh]

EXECUTE sofia/internal/306 at domain1.com set(call_timeout=30)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [call_timeout]=[30]

EXECUTE sofia/internal/306 at domain1.com set(hangup_after_bridge=true)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [hangup_after_bridge]=[true]

EXECUTE sofia/internal/306 at domain1.com set(continue_on_fail=true)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [continue_on_fail]=[true]

EXECUTE sofia/internal/306 at domain1.com
hash(insert/domain1.com-call_return/201/306)

EXECUTE sofia/internal/306 at domain1.com
hash(insert/domain1.com-last_dial_ext/201/b62db898-d6b1-11e1-be15-73f389528c
7f)

EXECUTE sofia/internal/306 at domain1.com set(called_party_callgroup=)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [called_party_callgroup]=[UNDEF]

EXECUTE sofia/internal/306 at domain1.com
hash(insert/domain1.com-last_dial//b62db898-d6b1-11e1-be15-73f389528c7f)

EXECUTE sofia/internal/306 at domain1.com bridge(user/201 at domain1.com)

2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1047
sofia/internal/306 at domain1.com EXPORTING[export_vars]
[domain_name]=[domain1.com] to event

2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1047
sofia/internal/306 at domain1.com EXPORTING[export_vars]
[dialed_extension]=[201] to event

2012-07-26 09:37:35.178096 [DEBUG] switch_ivr_originate.c:1958 Parsing
global variables

2012-07-26 09:37:35.198097 [DEBUG] switch_channel.c:1047
sofia/internal/306 at domain1.com EXPORTING[export_vars]
[domain_name]=[domain1.com] to event

2012-07-26 09:37:35.198097 [DEBUG] switch_channel.c:1047
sofia/internal/306 at domain1.com EXPORTING[export_vars]
[dialed_extension]=[201] to event

2012-07-26 09:37:35.198097 [DEBUG] switch_ivr_originate.c:1958 Parsing
global variables

2012-07-26 09:37:35.198097 [DEBUG] switch_event.c:1470 Parsing variable
[sip_invite_domain]=[domain1.com]

2012-07-26 09:37:35.198097 [DEBUG] switch_event.c:1470 Parsing variable
[presence_id]=[201 at domain1.com]

2012-07-26 09:37:35.198097 [NOTICE] switch_channel.c:926 New Channel
sofia/internal/sip:201 at 110.143.31.101:1178
[b630218c-d6b1-11e1-be1d-73f389528c7f]

2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:4734
(sofia/internal/sip:201 at 110.143.31.101:1178) State Change CS_NEW -> CS_INIT

2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:1228 Send signal
sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK]

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/sip:201 at 110.143.31.101:1178) Running State Change CS_INIT

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/sip:201 at 110.143.31.101:1178) State INIT

2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:85
sofia/internal/sip:201 at 110.143.31.101:1178 SOFIA INIT

2012-07-26 09:37:35.198097 [DEBUG] sofia_glue.c:2602 Local SDP:

v=0

o=FreeSWITCH 1343234313 1343234314 IN IP4 203.174.163.226

s=FreeSWITCH

c=IN IP4 203.174.163.226

t=0 0

m=audio 25142 RTP/AVP 8 0 3 101 13

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:125
(sofia/internal/sip:201 at 110.143.31.101:1178) State Change CS_INIT ->
CS_ROUTING

2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:1228 Send signal
sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK]

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/sip:201 at 110.143.31.101:1178) State INIT going to sleep

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/sip:201 at 110.143.31.101:1178) Running State Change CS_ROUTING

2012-07-26 09:37:35.198097 [DEBUG] switch_channel.c:1919
(sofia/internal/sip:201 at 110.143.31.101:1178) Callstate Change DOWN ->
RINGING

2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:923 Send signal
sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK]

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:433
(sofia/internal/sip:201 at 110.143.31.101:1178) State ROUTING

2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:148
sofia/internal/sip:201 at 110.143.31.101:1178 SOFIA ROUTING

2012-07-26 09:37:35.198097 [DEBUG] switch_ivr_originate.c:67
(sofia/internal/sip:201 at 110.143.31.101:1178) State Change CS_ROUTING ->
CS_CONSUME_MEDIA

2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:1228 Send signal
sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK]

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:433
(sofia/internal/sip:201 at 110.143.31.101:1178) State ROUTING going to sleep

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/sip:201 at 110.143.31.101:1178) Running State Change
CS_CONSUME_MEDIA

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:452
(sofia/internal/sip:201 at 110.143.31.101:1178) State CONSUME_MEDIA

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:452
(sofia/internal/sip:201 at 110.143.31.101:1178) State CONSUME_MEDIA going to
sleep

2012-07-26 09:37:35.198097 [DEBUG] sofia.c:5838 Channel
sofia/internal/sip:201 at 110.143.31.101:1178 entering state [calling][0]

2012-07-26 09:37:35.318096 [DEBUG] switch_core_session.c:923 Send signal
sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK]

2012-07-26 09:37:35.318096 [DEBUG] switch_core_session.c:923 Send signal
sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK]

2012-07-26 09:37:35.338097 [DEBUG] sofia.c:5838 Channel
sofia/internal/sip:201 at 110.143.31.101:1178 entering state [proceeding][180]

2012-07-26 09:37:35.338097 [NOTICE] sofia.c:5930 Ring-Ready
sofia/internal/sip:201 at 110.143.31.101:1178!

2012-07-26 09:37:35.338097 [INFO] switch_ivr_originate.c:1156 Sending early
media

2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3269 AUDIO RTP
[sofia/internal/306 at domain1.com] 203.174.163.226 port 26698 -> 115.64.93.203
port 55938 codec: 8 ms: 20

2012-07-26 09:37:35.338097 [DEBUG] switch_rtp.c:1680 Starting timer [soft]
160 bytes per 20ms

2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to
60ms (3 frames)

2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send
payload to 101

2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive
payload to 101

2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3566
sofia/internal/306 at domain1.com Set rtp dtmf delay to 40

2012-07-26 09:37:35.338097 [DEBUG] mod_sofia.c:2606 Ring SDP:

v=0

o=FreeSWITCH 1343232757 1343232758 IN IP4 203.174.163.226

s=FreeSWITCH

c=IN IP4 203.174.163.226

t=0 0

m=audio 26698 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

 

2012-07-26 09:37:35.338097 [NOTICE] mod_sofia.c:2609 Pre-Answer
sofia/internal/306 at domain1.com!

2012-07-26 09:37:35.338097 [DEBUG] switch_channel.c:3042
(sofia/internal/306 at domain1.com) Callstate Change RINGING -> EARLY

2012-07-26 09:37:35.338097 [DEBUG] switch_core_session.c:777 Send signal
sofia/internal/306 at domain1.com [BREAK]

 

-------------- next part --------------
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From mitch.capper at gmail.com  Wed Aug  1 18:57:00 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 1 Aug 2012 07:57:00 -0700
Subject: [Freeswitch-users] Help!! FS -TLS interworking issue,
 How to config to allow "gentls_cert" to generate a root certificate
 with more longer valid-period ?
In-Reply-To: <CAKzWOxVhuSy-G14TOUwFF3SHVxDOuMpVC5mnrS6VkuAA+pcWHg@mail.gmail.com>
References: <CAOTmbMxWPsgKXsXXDm1YSS1cUvsJvouSjq2mKjT_VvDShDQs5A@mail.gmail.com>
	<CALn233otcWsUJ8H0Qm+KrVDW6PbW-3waLhz7cmophBw3=8trXA@mail.gmail.com>
	<CAMWiMy-BMHovtqP=kOkVgm2b7MrWNXzofEsq8CrnOKT-4T0KFA@mail.gmail.com>
	<CAOTmbMzNNOzWvN29wuZ3wcV++ZB0odqWoeqk0nwe5sAiVF2-2Q@mail.gmail.com>
	<71943DD5C22943448A24B7C5CDC238070FF1A549@SN2PRD0410MB396.namprd04.prod.outlook.com>
	<CAKzWOxVhuSy-G14TOUwFF3SHVxDOuMpVC5mnrS6VkuAA+pcWHg@mail.gmail.com>
Message-ID: <CALn233q3BJL2ok-PiNsDXGQEBz-8EtUwrduZgtMGG89-re61EA@mail.gmail.com>

The changes to the attached genttls script verse headis the DAYS was
set to 365 instead of 2190 (6 years).   In addition the days variable
was quoted in two places it was not quoted.   6 years should not be
the cause of any problems so we are left with the quoting.   I tested
across 4 platforms without finding the quoting to be an issue.

Robert can you let us know what platform has an issue with the days
param not being quoted for days = 2190?

~mitch

On Mon, Jul 30, 2012 at 10:07 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Mon, Jul 30, 2012 at 9:28 AM, Robert Hadley <robert.hadley at teotech.com>
> wrote:
>>
>> Hi Charles,
>>
>>
>>
>> Try the changes in this attached freeswitch/scripts/gentls_cert.in file.
>> There were a few typos in the original script.
>>
>>
>>
>> Regards,
>>
>> Robert
>
>
> I'd like to verify that those typos are indeed really typos and are really
> fixed. If anyone has input on them please let me know and I will see about
> getting the gentls_cert.in file updated. I definitely would like to see this
> tested before we make any updates.
>
> Thanks,
> MC
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From evgeniy at bestnet.kharkov.ua  Wed Aug  1 19:00:38 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Wed, 01 Aug 2012 18:00:38 +0300
Subject: [Freeswitch-users] Nibble Billing deducted balance from account
 even call doesn't come to my mobile number
In-Reply-To: <CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
Message-ID: <50194496.10804@bestnet.kharkov.ua>

It did not work for me, i got the same message.
My outbond extension:

<extension name="AUTH_rad">
  <condition field="${toll_allow}" expression="international"/>
   <condition field="${AUTH_RESULT_RADIUS}" expression="^NOK$" 
break="on-true">
    <action application="lua" data="callfailed.lua ${caller_id_number}"/>
   </condition>
   <condition field="${AUTH_RESULT_RADIUS}" expression="^OK$">
    <action application="set" 
data="effective_caller_id_number=${outbound_caller_id_number}"/>
    <action application="set" 
data="effective_caller_id_name=${outbound_caller_id_name}"/>
    <action application="set" data="ignore_early_media=true"/>
    <action application="bridge" 
data="sofia/gateway/${default_gateway}/${dialed_number}"/>
   </condition>
</extension>


31.07.2012 11:06, SamyGo ?????:
> Hi,
> Well its works perfect for me, do you guys have ignore_early_media set in
> your outbound string, if no then set it and then see what happens.
> On Jul 31, 2012 12:50 PM, "Evgeniy Movlyan"<evgeniy at bestnet.kharkov.ua>
> wrote:
>
>> I have the same problem. When i am calling from one my extension to
>> another all is ok, but when i am calling to external number i got this
>> message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is not in
>> answered state".
>>
>> 31.07.2012 10:35, virendra bhati ?????:
>>> mod_nibblebill.c:465 Not billing
>>> 97183008 - call is not in answered state
>>
>> --
>> Evgeniy Movlyan,
>> BestNet Ltd.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Evgeniy Movlyan,
BestNet Ltd.


From jerry.richards at teotech.com  Wed Aug  1 19:12:33 2012
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 1 Aug 2012 15:12:33 +0000
Subject: [Freeswitch-users] Help!! FS -TLS interworking issue,
 How to config to allow "gentls_cert" to generate a root certificate
 with more longer valid-period ?
In-Reply-To: <CALn233q3BJL2ok-PiNsDXGQEBz-8EtUwrduZgtMGG89-re61EA@mail.gmail.com>
References: <CAOTmbMxWPsgKXsXXDm1YSS1cUvsJvouSjq2mKjT_VvDShDQs5A@mail.gmail.com>
	<CALn233otcWsUJ8H0Qm+KrVDW6PbW-3waLhz7cmophBw3=8trXA@mail.gmail.com>
	<CAMWiMy-BMHovtqP=kOkVgm2b7MrWNXzofEsq8CrnOKT-4T0KFA@mail.gmail.com>
	<CAOTmbMzNNOzWvN29wuZ3wcV++ZB0odqWoeqk0nwe5sAiVF2-2Q@mail.gmail.com>
	<71943DD5C22943448A24B7C5CDC238070FF1A549@SN2PRD0410MB396.namprd04.prod.outlook.com>
	<CAKzWOxVhuSy-G14TOUwFF3SHVxDOuMpVC5mnrS6VkuAA+pcWHg@mail.gmail.com>
	<CALn233q3BJL2ok-PiNsDXGQEBz-8EtUwrduZgtMGG89-re61EA@mail.gmail.com>
Message-ID: <1545146083A72C4DB7B66584B7E5D9841D10E5E7@BY2PRD0410MB377.namprd04.prod.outlook.com>

The platform Robert and I are using is CentOS 5.7.

Jerry

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch Capper
Sent: Wednesday, August 01, 2012 7:57 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Help!! FS -TLS interworking issue, How to config to allow "gentls_cert" to generate a root certificate with more longer valid-period ?

The changes to the attached genttls script verse headis the DAYS was
set to 365 instead of 2190 (6 years).   In addition the days variable
was quoted in two places it was not quoted.   6 years should not be
the cause of any problems so we are left with the quoting.   I tested
across 4 platforms without finding the quoting to be an issue.

Robert can you let us know what platform has an issue with the days param not being quoted for days = 2190?

~mitch

On Mon, Jul 30, 2012 at 10:07 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Mon, Jul 30, 2012 at 9:28 AM, Robert Hadley 
> <robert.hadley at teotech.com>
> wrote:
>>
>> Hi Charles,
>>
>>
>>
>> Try the changes in this attached freeswitch/scripts/gentls_cert.in file.
>> There were a few typos in the original script.
>>
>>
>>
>> Regards,
>>
>> Robert
>
>
> I'd like to verify that those typos are indeed really typos and are 
> really fixed. If anyone has input on them please let me know and I 
> will see about getting the gentls_cert.in file updated. I definitely 
> would like to see this tested before we make any updates.
>
> Thanks,
> MC
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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




From andrew at cassidywebservices.co.uk  Wed Aug  1 19:27:57 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 1 Aug 2012 16:27:57 +0100
Subject: [Freeswitch-users] FreeSWITCH dying with no core file
In-Reply-To: <CAE2A9TDGczBNAss75AE5pVqMutotrkec7uvPWmJwjkV+t8+AZA@mail.gmail.com>
References: <CAE2A9TCfrCw1=9ScVzsCcnP0CvaYVpMbHPAm19ChwY3rEQrN8A@mail.gmail.com>
	<5013012C.9050003@freeswitch.org>
	<CAE2A9TDGczBNAss75AE5pVqMutotrkec7uvPWmJwjkV+t8+AZA@mail.gmail.com>
Message-ID: <CAMbW9oKSFLCF-32rOD1g2wA4C_0s2DL-mSd8JmqWRbyTA+ed=Q@mail.gmail.com>

Hi, what have you done with the core file?

On 31 July 2012 21:16, Adam Johnston <ajohnston at blimessaging.com> wrote:

> That did it. My sporadic crash happened again and, sure enough, there's a
> core file.
>
> Thanks again,
> Adam Johnston
>
>
> On Fri, Jul 27, 2012 at 4:59 PM, Stefan Knoblich <stkn at freeswitch.org>wrote:
>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> On 07/27/12 21:34, Adam Johnston wrote:
>> > I use FreeSWITCH exclusively for high-volume fax, both sending and
>> receiving, and I've been running into a sporadic issue where the FreeSWITCH
>> process will die with nothing in the logs to
>> > indicate an issue and no core file anywhere on the system. I'm setting
>> "ulimit -c unlimited" in my init script, so I would expect to see a core
>> file. Version: FreeSWITCH Version
>> > 1.2.0-rc2+git~20120713T162602Z~0afd7318bd+unclean~20120713T182401Z
>>
>> You need to set /proc/sys/fs/suid_dumpable to a non-zero value (see man 5
>> proc), to enable core dumping for processes
>> that switch their UID on startup (such as freeswitch with -u option).
>>
>> By default, the freeswitch process will try to put the core file into the
>> CWD (current work directory),
>> so make sure the UID it is running as can write there (or "cd" into a
>> different directory before starting it).
>> You might also want to check out man 5 core, for all the other options
>> (e.g. piping core dumps into a custom handler, with core_pattern).
>>
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v2.0.19 (GNU/Linux)
>> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
>>
>> iEYEARECAAYFAlATASwACgkQjiIIAK4rYUrjvQCgh1grtcBfPVhbZNw8tdQpsyda
>> 614Amwb1pKDFL93dU33cy2wj1FNYYnzJ
>> =0Xqk
>> -----END PGP SIGNATURE-----
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
-------------- next part --------------
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From mitch.capper at gmail.com  Wed Aug  1 19:46:55 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 1 Aug 2012 08:46:55 -0700
Subject: [Freeswitch-users] Help!! FS -TLS interworking issue,
 How to config to allow "gentls_cert" to generate a root certificate
 with more longer valid-period ?
In-Reply-To: <1545146083A72C4DB7B66584B7E5D9841D10E5E7@BY2PRD0410MB377.namprd04.prod.outlook.com>
References: <CAOTmbMxWPsgKXsXXDm1YSS1cUvsJvouSjq2mKjT_VvDShDQs5A@mail.gmail.com>
	<CALn233otcWsUJ8H0Qm+KrVDW6PbW-3waLhz7cmophBw3=8trXA@mail.gmail.com>
	<CAMWiMy-BMHovtqP=kOkVgm2b7MrWNXzofEsq8CrnOKT-4T0KFA@mail.gmail.com>
	<CAOTmbMzNNOzWvN29wuZ3wcV++ZB0odqWoeqk0nwe5sAiVF2-2Q@mail.gmail.com>
	<71943DD5C22943448A24B7C5CDC238070FF1A549@SN2PRD0410MB396.namprd04.prod.outlook.com>
	<CAKzWOxVhuSy-G14TOUwFF3SHVxDOuMpVC5mnrS6VkuAA+pcWHg@mail.gmail.com>
	<CALn233q3BJL2ok-PiNsDXGQEBz-8EtUwrduZgtMGG89-re61EA@mail.gmail.com>
	<1545146083A72C4DB7B66584B7E5D9841D10E5E7@BY2PRD0410MB377.namprd04.prod.outlook.com>
Message-ID: <CALn233rZUS=8yPczpTPBsxGBEe8QQFSDDaZLo=aB84vHesOmBQ@mail.gmail.com>

Is anyone else able to confirm a problem on CentOS 5.7 with the
genttls_cert from head?    I have tried it on CentOS 6 and some pre 5
fedora boxes that all seem to work correctly but I do not have any
running CentOS 5.  To test run:
./gentls_cert setup && ./gentls_cert create_server
openssl x509 -noout -in /usr/local/freeswitch/conf/ssl/agent.pem  -enddate
openssl x509 -noout -in /usr/local/freeswitch/conf/ssl/cafile.pem  -enddate

The date should be in 2018.

if you get any errors please let us know the error.

~Mitch

On Wed, Aug 1, 2012 at 8:12 AM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> The platform Robert and I are using is CentOS 5.7.
>
> Jerry
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch Capper
> Sent: Wednesday, August 01, 2012 7:57 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Help!! FS -TLS interworking issue, How to config to allow "gentls_cert" to generate a root certificate with more longer valid-period ?
>
> The changes to the attached genttls script verse headis the DAYS was
> set to 365 instead of 2190 (6 years).   In addition the days variable
> was quoted in two places it was not quoted.   6 years should not be
> the cause of any problems so we are left with the quoting.   I tested
> across 4 platforms without finding the quoting to be an issue.
>
> Robert can you let us know what platform has an issue with the days param not being quoted for days = 2190?
>
> ~mitch
>
> On Mon, Jul 30, 2012 at 10:07 AM, Michael Collins <msc at freeswitch.org> wrote:
>>
>>
>> On Mon, Jul 30, 2012 at 9:28 AM, Robert Hadley
>> <robert.hadley at teotech.com>
>> wrote:
>>>
>>> Hi Charles,
>>>
>>>
>>>
>>> Try the changes in this attached freeswitch/scripts/gentls_cert.in file.
>>> There were a few typos in the original script.
>>>
>>>
>>>
>>> Regards,
>>>
>>> Robert
>>
>>
>> I'd like to verify that those typos are indeed really typos and are
>> really fixed. If anyone has input on them please let me know and I
>> will see about getting the gentls_cert.in file updated. I definitely
>> would like to see this tested before we make any updates.
>>
>> Thanks,
>> MC
>>
>> ______________________________________________________________________
>> ___ Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From jerry.richards at teotech.com  Wed Aug  1 19:49:16 2012
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 1 Aug 2012 15:49:16 +0000
Subject: [Freeswitch-users] send_silence_when_idle=10000 Breaks
	inherit_codec?
Message-ID: <1545146083A72C4DB7B66584B7E5D9841D10E61D@BY2PRD0410MB377.namprd04.prod.outlook.com>

Hello All,

We want to configure Freeswitch with the following tags, but there are undesired side-effects.

In conf/vars.xml:
<X-PRE-PROCESS cmd="set" data="send_silence_when_idle=10000"/>

In conf/sip_profiles/internal.xml:
<param name="inbound-late-negotiation" value="true"/>

In conf/dialplan/default.xml:
<param name="inherit_codec" value="true"/>

The problem is that setting send_silence_when_idle in vars.xml (above) causes ringback_data to get set in switch_ivr_originate.c (around line 2070) which causes 183 Session Progress w/SDP which prevents inherit_codec in internal.xml (above) to work as expected.

This looks like a bug?

Thanks,
Jerry



From Hector.Geraldino at ipsoft.com  Wed Aug  1 19:52:06 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Wed, 1 Aug 2012 11:52:06 -0400
Subject: [Freeswitch-users] Am I using play_and_detect_speech correctly?
In-Reply-To: <D346F0E06A1D9341B7FF1FC1B525698C5C1291@ex10.mail.win.hw.ac.uk>
References: <D346F0E06A1D9341B7FF1FC1B525698C5C128A@ex10.mail.win.hw.ac.uk><CAD-qRkhF-=6y6xWK0eyuZN3FSCUw7qAhHwyHbnJnXTmoZxZXrA@mail.gmail.com><50179069.6020700@hw.ac.uk>
	<CAD-qRkiL-KFF-tkKk3Ni=eOn+A6dp5eKxe61PK=6_m5N_WBaFw@mail.gmail.com>
	<D346F0E06A1D9341B7FF1FC1B525698C5C1291@ex10.mail.win.hw.ac.uk>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD02304F6260@NY1-EXMB-01.ip-soft.net>

You need to send a detect_speech command with sendmsg + execute:

sendmsg <uuid>
call-command: execute
execute-app-name: detect_speech
execute-app-arg: grammar_name //e.g. unimrcp:nuance5-mrcp2-prod nuancegrm nuancegrm

This will trigger an DETECTED_SPEECH event (you need to add this event to your filters, in case you're filtering the events). The detected speech will be part of the bodyLines of the event. I'm using Java ESL client libraries (org.freeswitch.esl.client) and my code looks like:

protected void handleEslEvent(ChannelHandlerContext ctx, EslEvent event) {
    if (event.getEventName().equals("DETECTED_SPEECH")) {
        if (event.getEventBodyLines().size() > 0) {

            final String uuid = event.getEventHeaders().get("Unique-ID");
            stopSpeechDetection(uuid);

           //detected speech
            StringBuilder bodyLines = new StringBuilder();

            for (String line : event.getEventBodyLines()) {
                bodyLines.append(line);
            }
        }
    }
}

private void stopSpeechDetection(CallSession call) {
    // sendMessage(uuid, "detect_speech", "stop");
}

g.luck!

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Liu, Xingkun
Sent: Wednesday, August 01, 2012 4:49 AM
To: FreeSWITCH Users Help; FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Am I using play_and_detect_speech correctly?


Hi Chris,

Many thanks for your info and suggestions!

Would you please tell me a bit more or any clue about the idea of using detect_speech APP to handle
barge-in over ESL? It is an interesting idea to try but for now I don't have any clue how it could be
done over Java ESL without touching or modifying detect_speech APP codes.

Thanks again!

Best regards,

Xing




-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> on behalf of Christopher Rienzo
Sent: Tue 7/31/2012 13:44
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Am I using play_and_detect_speech correctly?

I looked over the code again last night and verified that
play_and_detect_speech should try to stop detection before loading the
grammar.  However, that scenario is not really tested, so try the
workaround first.  If that doesn't work, open a jira ticket with detailed
logs so it can be fixed.

You can use the detect_speech APP to do everything over ESL.  It is a lot
more work, but once you figure it out, you have the freedom to do whatever
you want, including handling barge-in.  play_and_detect_speech is designed
to handle the typical use case.

Chris



On Tue, Jul 31, 2012 at 3:59 AM, x.liu <x.liu at hw.ac.uk<mailto:x.liu at hw.ac.uk>> wrote:

>  Hi Chris,
>
> Okay, thanks! I will have tries to see how it works.
>
> For the normal use of the app: play_and_detect_speech, will it resume
> previous ASR session,
> or will it stop previous session and restart a new session each time when
> I issue it via ESL?
>
> For the app of detect_speech, I can first to start an ASR session by
> "detect_speech speechMod grammar grammarPath",
> then I can do "detect_speech pause", or "detect_speech resume" or
> "detect_speech stop".
> But for play_and_detect_speech, I am not sure how it works in terms of
> starting, pause, resume/restart sequence.
>
> The play_and_detect_speech is a very good, useful app as it supports the
> barge-in. So we definitely need to use it.
>
> Thanks!
>
> Xing
>
>
>
>
>
> On 07/31/2012 12:53 AM, Christopher Rienzo wrote:
>
> You are using that APP in a way that hasn't been tested.  Try to do
> "speech_detect pause" and see if that helps.  It will reserve your ASR
> session for reuse while "speech_detect stop" will tear it down completely.
>  If that doesn't work, open a jira ticket and attach a full call trace.
>  It's difficult to understand exactly what is happening from your
> description.
>
>
>  Chris
>
>
>
> On Mon, Jul 30, 2012 at 5:38 PM, Liu, Xingkun <X.Liu at hw.ac.uk<mailto:X.Liu at hw.ac.uk>> wrote:
>
>>  Hello,
>>
>> I am using play_and_detect_speech with Java ESL in my IVR applications.
>>
>> Previously I call it again each time after I receive any recognition
>> event,
>> like recognition complete, no-input-timeout, or recognition-timeout,
>> it seems to work fine.
>>
>> Now I have changed my app to issue play_and_detect_speech command based on
>> my available system utterances as well as the speech event.
>>
>> I.e., I use a separate thread to constantly check if there is a system
>> utterance coming in
>> from another component of my application, if there is any utterance I
>> issue the command which will
>> speak the new utterance and listen to user input no matter whether or not
>> previous
>> command has finished. And if there is any speech event (recognition
>> result, timeout etc.)
>> the play_and_detect_speech command is also issued but with playing
>> silence.
>>
>> Obviously the new command will stop the utterance speaking of the
>> previous command if it is not finished.
>>
>> My question is
>>
>> will the new play_and_detect_speech command also stop the previous ASR
>> listening
>> or will there be many ASR listening channel and sending speech data (or
>> silence) to ASR server?
>>
>> Do I need to explicitly issue a "stop" commnad before issuing a new
>> play_and_detect_speech?
>> If yes, how to do that, by "detect_speech stop"?
>>
>> Recently there is a network traffice problem (lots of connections /data
>> transportation to the ASR server machine)
>> when I am running my application. I am not sure if this is because of
>> other issues
>> or because of my new changes to the way of using play_and_detect_speech.
>>
>> Please any one could shed a light on this?
>>
>> Many thanks!
>>
>> Xing
>>
>>
>> ------------------------------
>>
>>  *Heriot-Watt University is the Sunday Times Scottish University of the
>> Year 2011-2012.*
>>
>>  We invite research leaders and ambitious early career researchers to
>> join us in leading and driving research in key inter-disciplinary themes.
>> Please see www.hw.ac.uk/researchleaders<http://www.hw.ac.uk/researchleaders> for further information and how
>> to apply.
>>
>>  Heriot-Watt University is a Scottish charity registered under charity
>> number SC000278.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-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
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users<mailto: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<http://lists.freeswitch.org/mailman/options/freeswitch-usershttp:/www.freeswitch.org>
>
>
>
> ------------------------------
>
>  *Heriot-Watt University is the Sunday Times Scottish University of the
> Year 2011-2012.*
>
>  We invite research leaders and ambitious early career researchers to join
> us in leading and driving research in key inter-disciplinary themes. Please
> see www.hw.ac.uk/researchleaders<http://www.hw.ac.uk/researchleaders> for further information and how to
> apply.
>
>  Heriot-Watt University is a Scottish charity registered under charity
> number SC000278.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-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
>
>

________________________________

Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012.

We invite research leaders and ambitious early career researchers to join us in leading and driving research in key inter-disciplinary themes. Please see www.hw.ac.uk/researchleaders<http://www.hw.ac.uk/researchleaders> for further information and how to apply.

Heriot-Watt University is a Scottish charity registered under charity number SC000278.
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From msc at freeswitch.org  Wed Aug  1 19:55:41 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Aug 2012 08:55:41 -0700
Subject: [Freeswitch-users] FreeSWITCH Community Conference Call
Message-ID: <CAKzWOxUbztqPVx+kwsFz_2aVu4=k=QcE2jr0x4O1JMczHLR3SQ@mail.gmail.com>

Hello all!

We'll be having a community discussion today. The agenda page is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_01

Talk to you soon!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From mitch.capper at gmail.com  Wed Aug  1 20:00:45 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 1 Aug 2012 09:00:45 -0700
Subject: [Freeswitch-users] Help!! FS -TLS interworking issue,
 How to config to allow "gentls_cert" to generate a root certificate
 with more longer valid-period ?
In-Reply-To: <CALn233rZUS=8yPczpTPBsxGBEe8QQFSDDaZLo=aB84vHesOmBQ@mail.gmail.com>
References: <CAOTmbMxWPsgKXsXXDm1YSS1cUvsJvouSjq2mKjT_VvDShDQs5A@mail.gmail.com>
	<CALn233otcWsUJ8H0Qm+KrVDW6PbW-3waLhz7cmophBw3=8trXA@mail.gmail.com>
	<CAMWiMy-BMHovtqP=kOkVgm2b7MrWNXzofEsq8CrnOKT-4T0KFA@mail.gmail.com>
	<CAOTmbMzNNOzWvN29wuZ3wcV++ZB0odqWoeqk0nwe5sAiVF2-2Q@mail.gmail.com>
	<71943DD5C22943448A24B7C5CDC238070FF1A549@SN2PRD0410MB396.namprd04.prod.outlook.com>
	<CAKzWOxVhuSy-G14TOUwFF3SHVxDOuMpVC5mnrS6VkuAA+pcWHg@mail.gmail.com>
	<CALn233q3BJL2ok-PiNsDXGQEBz-8EtUwrduZgtMGG89-re61EA@mail.gmail.com>
	<1545146083A72C4DB7B66584B7E5D9841D10E5E7@BY2PRD0410MB377.namprd04.prod.outlook.com>
	<CALn233rZUS=8yPczpTPBsxGBEe8QQFSDDaZLo=aB84vHesOmBQ@mail.gmail.com>
Message-ID: <CALn233pfjirWtyW7gt_cwO0UDfarO4UEe4cGGUEKgoanDbeb-Q@mail.gmail.com>

To make it easier I put up a version anyone can grab and it won't
trample your existing FS data (installs certs to /tmp/fs_test):
wget http://mitchcapper.com/gentls_cert && chmod +x gentls_cert
./gentls_cert setup && ./gentls_cert create_server
openssl x509 -noout -in /tmp/fs_test/agent.pem  -enddate
openssl x509 -noout -in /tmp/fs_test/cafile.pem  -enddate


Once done just rm /tmp/fs_test  and genttls_cert and there will be
nothing remaining from the test.
~Mitch



On Wed, Aug 1, 2012 at 8:46 AM, Mitch Capper <mitch.capper at gmail.com> wrote:
> Is anyone else able to confirm a problem on CentOS 5.7 with the
> genttls_cert from head?    I have tried it on CentOS 6 and some pre 5
> fedora boxes that all seem to work correctly but I do not have any
> running CentOS 5.  To test run:
> ./gentls_cert setup && ./gentls_cert create_server
> openssl x509 -noout -in /usr/local/freeswitch/conf/ssl/agent.pem  -enddate
> openssl x509 -noout -in /usr/local/freeswitch/conf/ssl/cafile.pem  -enddate
>
> The date should be in 2018.
>
> if you get any errors please let us know the error.
>
> ~Mitch
>
> On Wed, Aug 1, 2012 at 8:12 AM, Jerry Richards
> <jerry.richards at teotech.com> wrote:
>> The platform Robert and I are using is CentOS 5.7.
>>
>> Jerry
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch Capper
>> Sent: Wednesday, August 01, 2012 7:57 AM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Help!! FS -TLS interworking issue, How to config to allow "gentls_cert" to generate a root certificate with more longer valid-period ?
>>
>> The changes to the attached genttls script verse headis the DAYS was
>> set to 365 instead of 2190 (6 years).   In addition the days variable
>> was quoted in two places it was not quoted.   6 years should not be
>> the cause of any problems so we are left with the quoting.   I tested
>> across 4 platforms without finding the quoting to be an issue.
>>
>> Robert can you let us know what platform has an issue with the days param not being quoted for days = 2190?
>>
>> ~mitch
>>
>> On Mon, Jul 30, 2012 at 10:07 AM, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>
>>> On Mon, Jul 30, 2012 at 9:28 AM, Robert Hadley
>>> <robert.hadley at teotech.com>
>>> wrote:
>>>>
>>>> Hi Charles,
>>>>
>>>>
>>>>
>>>> Try the changes in this attached freeswitch/scripts/gentls_cert.in file.
>>>> There were a few typos in the original script.
>>>>
>>>>
>>>>
>>>> Regards,
>>>>
>>>> Robert
>>>
>>>
>>> I'd like to verify that those typos are indeed really typos and are
>>> really fixed. If anyone has input on them please let me know and I
>>> will see about getting the gentls_cert.in file updated. I definitely
>>> would like to see this tested before we make any updates.
>>>
>>> Thanks,
>>> MC
>>>
>>> ______________________________________________________________________
>>> ___ Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>> rs
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>  
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org


From ajohnston at blimessaging.com  Wed Aug  1 20:04:34 2012
From: ajohnston at blimessaging.com (Adam Johnston)
Date: Wed, 1 Aug 2012 12:04:34 -0400
Subject: [Freeswitch-users] FreeSWITCH dying with no core file
In-Reply-To: <CAMbW9oKSFLCF-32rOD1g2wA4C_0s2DL-mSd8JmqWRbyTA+ed=Q@mail.gmail.com>
References: <CAE2A9TCfrCw1=9ScVzsCcnP0CvaYVpMbHPAm19ChwY3rEQrN8A@mail.gmail.com>
	<5013012C.9050003@freeswitch.org>
	<CAE2A9TDGczBNAss75AE5pVqMutotrkec7uvPWmJwjkV+t8+AZA@mail.gmail.com>
	<CAMbW9oKSFLCF-32rOD1g2wA4C_0s2DL-mSd8JmqWRbyTA+ed=Q@mail.gmail.com>
Message-ID: <CAE2A9TCosukvTHHa6YWWYskmNwFH8fheBQjceGPWpPy6r_mf_Q@mail.gmail.com>

Andrew,

I loaded the core file into gdb and looked at the backtrace. It turns out
that the problem was not in the FreeSWITCH code but rather in LuaSocket,
and the error was thrown while I was caching fax results in a key-value
store using a Lua script. My issue was very similar to this Jira:
http://jira.freeswitch.org/browse/FS-2893. I recompiled LuaSocket with the
FreeSWITCH includes and am hoping that fixes things.

Adam


On Wed, Aug 1, 2012 at 11:27 AM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> Hi, what have you done with the core file?
>
> On 31 July 2012 21:16, Adam Johnston <ajohnston at blimessaging.com> wrote:
>
>> That did it. My sporadic crash happened again and, sure enough, there's a
>> core file.
>>
>> Thanks again,
>> Adam Johnston
>>
>>
>> On Fri, Jul 27, 2012 at 4:59 PM, Stefan Knoblich <stkn at freeswitch.org>wrote:
>>
>>> -----BEGIN PGP SIGNED MESSAGE-----
>>> Hash: SHA1
>>>
>>> On 07/27/12 21:34, Adam Johnston wrote:
>>> > I use FreeSWITCH exclusively for high-volume fax, both sending and
>>> receiving, and I've been running into a sporadic issue where the FreeSWITCH
>>> process will die with nothing in the logs to
>>> > indicate an issue and no core file anywhere on the system. I'm setting
>>> "ulimit -c unlimited" in my init script, so I would expect to see a core
>>> file. Version: FreeSWITCH Version
>>> > 1.2.0-rc2+git~20120713T162602Z~0afd7318bd+unclean~20120713T182401Z
>>>
>>> You need to set /proc/sys/fs/suid_dumpable to a non-zero value (see man
>>> 5 proc), to enable core dumping for processes
>>> that switch their UID on startup (such as freeswitch with -u option).
>>>
>>> By default, the freeswitch process will try to put the core file into
>>> the CWD (current work directory),
>>> so make sure the UID it is running as can write there (or "cd" into a
>>> different directory before starting it).
>>> You might also want to check out man 5 core, for all the other options
>>> (e.g. piping core dumps into a custom handler, with core_pattern).
>>>
>>> -----BEGIN PGP SIGNATURE-----
>>> Version: GnuPG v2.0.19 (GNU/Linux)
>>> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
>>>
>>> iEYEARECAAYFAlATASwACgkQjiIIAK4rYUrjvQCgh1grtcBfPVhbZNw8tdQpsyda
>>> 614Amwb1pKDFL93dU33cy2wj1FNYYnzJ
>>> =0Xqk
>>> -----END PGP SIGNATURE-----
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Aug  1 20:13:58 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Aug 2012 09:13:58 -0700
Subject: [Freeswitch-users] send_silence_when_idle=10000 Breaks
	inherit_codec?
In-Reply-To: <1545146083A72C4DB7B66584B7E5D9841D10E61D@BY2PRD0410MB377.namprd04.prod.outlook.com>
References: <1545146083A72C4DB7B66584B7E5D9841D10E61D@BY2PRD0410MB377.namprd04.prod.outlook.com>
Message-ID: <CAKzWOxW35ec12ygCCD5AyO4Nob4B2L2FaXz9Aj2h8rUyHMtHAg@mail.gmail.com>

Yep, get that in a Jira, STAT!
-MC

On Wed, Aug 1, 2012 at 8:49 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> Hello All,
>
> We want to configure Freeswitch with the following tags, but there are
> undesired side-effects.
>
> In conf/vars.xml:
> <X-PRE-PROCESS cmd="set" data="send_silence_when_idle=10000"/>
>
> In conf/sip_profiles/internal.xml:
> <param name="inbound-late-negotiation" value="true"/>
>
> In conf/dialplan/default.xml:
> <param name="inherit_codec" value="true"/>
>
> The problem is that setting send_silence_when_idle in vars.xml (above)
> causes ringback_data to get set in switch_ivr_originate.c (around line
> 2070) which causes 183 Session Progress w/SDP which prevents inherit_codec
> in internal.xml (above) to work as expected.
>
> This looks like a bug?
>
> Thanks,
> Jerry
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Aug  1 20:18:32 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 1 Aug 2012 09:18:32 -0700
Subject: [Freeswitch-users] ESL - phpmod compile Error
In-Reply-To: <CAGvuWYa7c7Fy+B8gKNU4cSFLT6HtjYpC_03UvqezRRJSw4dXhw@mail.gmail.com>
References: <CAGvuWYa7c7Fy+B8gKNU4cSFLT6HtjYpC_03UvqezRRJSw4dXhw@mail.gmail.com>
Message-ID: <CAKzWOxVYu2HYAJ2X=jAdbJW=dZmZAniNBxtmNhq6057FuXCdBQ@mail.gmail.com>

I suspect this line is the key piece of information:
/usr/bin/ld: cannot find -ledit

If I read that correctly you need to install libedit-dev. Give that a try
and let us know.

-MC

On Tue, Jul 31, 2012 at 11:20 PM, Lloyd Aloysius
<lloyd.aloysius at gmail.com>wrote:

> Hi All:
>
> when I compile the ESL module for php . OS - Centos 6.2 / 64 bit
>
> libs/esl
>
> make phpmod
>
> I got the following Errors.
>
>   make phpmod
> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
> CFLAGS="-I/usr/src/freeswitch.git/libs/esl/src/include -DHAVE_EDITLINE -g
> -ggdb -I../../libs/libedit/src/ -fPIC -O2"
> CXXFLAGS="-I/usr/src/freeswitch.git/libs/esl/src/include -DHAVE_EDITLINE -g
> -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php
> make[1]: Entering directory `/usr/src/freeswitch.git/libs/esl/php'
> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib6464 -lcrypt
> -lcrypt -ledit -lncurses -lm -ldl -lnsl -lm -ldl -ldl -lm -lcrypt -lm
> -lcrypt -lpthread -o ESL.so -L.
> /usr/bin/ld: cannot find -ledit
> collect2: ld returned 1 exit status
> make[1]: *** [ESL.so] Error 1
> make[1]: Leaving directory `/usr/src/freeswitch.git/libs/esl/php'
> make: *** [phpmod] Error 2
>
>
> How to solve this issue.
>
> Thanks
> Lloyd
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From freeswitch-list at puzzled.xs4all.nl  Wed Aug  1 20:56:28 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Wed, 01 Aug 2012 18:56:28 +0200
Subject: [Freeswitch-users] Help!! FS -TLS interworking issue,
 How to config to allow "gentls_cert" to generate a root certificate
 with more longer valid-period ?
In-Reply-To: <CALn233pfjirWtyW7gt_cwO0UDfarO4UEe4cGGUEKgoanDbeb-Q@mail.gmail.com>
References: <CAOTmbMxWPsgKXsXXDm1YSS1cUvsJvouSjq2mKjT_VvDShDQs5A@mail.gmail.com>
	<CALn233otcWsUJ8H0Qm+KrVDW6PbW-3waLhz7cmophBw3=8trXA@mail.gmail.com>
	<CAMWiMy-BMHovtqP=kOkVgm2b7MrWNXzofEsq8CrnOKT-4T0KFA@mail.gmail.com>
	<CAOTmbMzNNOzWvN29wuZ3wcV++ZB0odqWoeqk0nwe5sAiVF2-2Q@mail.gmail.com>
	<71943DD5C22943448A24B7C5CDC238070FF1A549@SN2PRD0410MB396.namprd04.prod.outlook.com>
	<CAKzWOxVhuSy-G14TOUwFF3SHVxDOuMpVC5mnrS6VkuAA+pcWHg@mail.gmail.com>
	<CALn233q3BJL2ok-PiNsDXGQEBz-8EtUwrduZgtMGG89-re61EA@mail.gmail.com>
	<1545146083A72C4DB7B66584B7E5D9841D10E5E7@BY2PRD0410MB377.namprd04.prod.outlook.com>
	<CALn233rZUS=8yPczpTPBsxGBEe8QQFSDDaZLo=aB84vHesOmBQ@mail.gmail.com>
	<CALn233pfjirWtyW7gt_cwO0UDfarO4UEe4cGGUEKgoanDbeb-Q@mail.gmail.com>
Message-ID: <50195FBC.4090102@puzzled.xs4all.nl>

On 01-08-12 18:00, Mitch Capper wrote:
> To make it easier I put up a version anyone can grab and it won't
> trample your existing FS data (installs certs to /tmp/fs_test):
> wget http://mitchcapper.com/gentls_cert && chmod +x gentls_cert
> ./gentls_cert setup && ./gentls_cert create_server
> openssl x509 -noout -in /tmp/fs_test/agent.pem  -enddate
> openssl x509 -noout -in /tmp/fs_test/cafile.pem  -enddate
>
>
> Once done just rm /tmp/fs_test  and genttls_cert and there will be
> nothing remaining from the test.
> ~Mitch

I don't have 5.7 (why would one not update to 5.8?!) but on an 
up-to-date CentOS 5.8 x86 box I see the following from your script:

[patrick at svr2 fs_test]$ openssl x509 -noout -in agent.pem -enddate
notAfter=Jul 31 16:48:58 2018 GMT
[patrick at svr2 fs_test]$ openssl x509 -noout -in cafile.pem -enddate
notAfter=Jul 31 16:48:57 2018 GMT

Regards,
Patrick


From X.Liu at hw.ac.uk  Wed Aug  1 21:49:29 2012
From: X.Liu at hw.ac.uk (Liu, Xingkun)
Date: Wed, 1 Aug 2012 18:49:29 +0100
Subject: [Freeswitch-users] Am I using play_and_detect_speech correctly?
References: <D346F0E06A1D9341B7FF1FC1B525698C5C128A@ex10.mail.win.hw.ac.uk><CAD-qRkhF-=6y6xWK0eyuZN3FSCUw7qAhHwyHbnJnXTmoZxZXrA@mail.gmail.com><50179069.6020700@hw.ac.uk><CAD-qRkiL-KFF-tkKk3Ni=eOn+A6dp5eKxe61PK=6_m5N_WBaFw@mail.gmail.com><D346F0E06A1D9341B7FF1FC1B525698C5C1291@ex10.mail.win.hw.ac.uk>
	<6A6B4C284AD15042B429EB9D904544AD02304F6260@NY1-EXMB-01.ip-soft.net>
Message-ID: <D346F0E06A1D9341B7FF1FC1B525698C5C1299@ex10.mail.win.hw.ac.uk>

Thanks for your response, Hector!

Yeah, I am using detect_speech via a similar way to yours.

What I am more interested in is to use detect_speech app to handle user's barge-in.

After Chris mentioned the barge-in can be also handled by detect_speech I gave it a further thinking.
Yeah, I could first "speak" the utterance and immediately resume ASR, then try to catch the begin_speaking event, 
then stop the TTS -- using this way to handle the user barge-in.
(Chris, you may have a better idea, would you please let me know if you do?)

One thing I am worry about is that stopping currently playing media or utterance seems not work for me.
When I recently try "api uuid_break <uuid>" it stopped currently playing music but also stopped playing following utterances
which I sent to TTS soon later on after uuid_break.

Anyway I will try it further again and let you all know what I will get.

Cheers,

Xing


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Hector Geraldino
Sent: Wed 8/1/2012 16:52
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Am I using play_and_detect_speech correctly?
 
You need to send a detect_speech command with sendmsg + execute:

sendmsg <uuid>
call-command: execute
execute-app-name: detect_speech
execute-app-arg: grammar_name //e.g. unimrcp:nuance5-mrcp2-prod nuancegrm nuancegrm

This will trigger an DETECTED_SPEECH event (you need to add this event to your filters, in case you're filtering the events). The detected speech will be part of the bodyLines of the event. I'm using Java ESL client libraries (org.freeswitch.esl.client) and my code looks like:

protected void handleEslEvent(ChannelHandlerContext ctx, EslEvent event) {
    if (event.getEventName().equals("DETECTED_SPEECH")) {
        if (event.getEventBodyLines().size() > 0) {

            final String uuid = event.getEventHeaders().get("Unique-ID");
            stopSpeechDetection(uuid);

           //detected speech
            StringBuilder bodyLines = new StringBuilder();

            for (String line : event.getEventBodyLines()) {
                bodyLines.append(line);
            }
        }
    }
}

private void stopSpeechDetection(CallSession call) {
    // sendMessage(uuid, "detect_speech", "stop");
}

g.luck!

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Liu, Xingkun
Sent: Wednesday, August 01, 2012 4:49 AM
To: FreeSWITCH Users Help; FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Am I using play_and_detect_speech correctly?


Hi Chris,

Many thanks for your info and suggestions!

Would you please tell me a bit more or any clue about the idea of using detect_speech APP to handle
barge-in over ESL? It is an interesting idea to try but for now I don't have any clue how it could be
done over Java ESL without touching or modifying detect_speech APP codes.

Thanks again!

Best regards,

Xing




-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> on behalf of Christopher Rienzo
Sent: Tue 7/31/2012 13:44
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Am I using play_and_detect_speech correctly?

I looked over the code again last night and verified that
play_and_detect_speech should try to stop detection before loading the
grammar.  However, that scenario is not really tested, so try the
workaround first.  If that doesn't work, open a jira ticket with detailed
logs so it can be fixed.

You can use the detect_speech APP to do everything over ESL.  It is a lot
more work, but once you figure it out, you have the freedom to do whatever
you want, including handling barge-in.  play_and_detect_speech is designed
to handle the typical use case.

Chris



On Tue, Jul 31, 2012 at 3:59 AM, x.liu <x.liu at hw.ac.uk<mailto:x.liu at hw.ac.uk>> wrote:

>  Hi Chris,
>
> Okay, thanks! I will have tries to see how it works.
>
> For the normal use of the app: play_and_detect_speech, will it resume
> previous ASR session,
> or will it stop previous session and restart a new session each time when
> I issue it via ESL?
>
> For the app of detect_speech, I can first to start an ASR session by
> "detect_speech speechMod grammar grammarPath",
> then I can do "detect_speech pause", or "detect_speech resume" or
> "detect_speech stop".
> But for play_and_detect_speech, I am not sure how it works in terms of
> starting, pause, resume/restart sequence.
>
> The play_and_detect_speech is a very good, useful app as it supports the
> barge-in. So we definitely need to use it.
>
> Thanks!
>
> Xing
>
>
>
>
>
> On 07/31/2012 12:53 AM, Christopher Rienzo wrote:
>
> You are using that APP in a way that hasn't been tested.  Try to do
> "speech_detect pause" and see if that helps.  It will reserve your ASR
> session for reuse while "speech_detect stop" will tear it down completely.
>  If that doesn't work, open a jira ticket and attach a full call trace.
>  It's difficult to understand exactly what is happening from your
> description.
>
>
>  Chris
>
>
>
> On Mon, Jul 30, 2012 at 5:38 PM, Liu, Xingkun <X.Liu at hw.ac.uk<mailto:X.Liu at hw.ac.uk>> wrote:
>
>>  Hello,
>>
>> I am using play_and_detect_speech with Java ESL in my IVR applications.
>>
>> Previously I call it again each time after I receive any recognition
>> event,
>> like recognition complete, no-input-timeout, or recognition-timeout,
>> it seems to work fine.
>>
>> Now I have changed my app to issue play_and_detect_speech command based on
>> my available system utterances as well as the speech event.
>>
>> I.e., I use a separate thread to constantly check if there is a system
>> utterance coming in
>> from another component of my application, if there is any utterance I
>> issue the command which will
>> speak the new utterance and listen to user input no matter whether or not
>> previous
>> command has finished. And if there is any speech event (recognition
>> result, timeout etc.)
>> the play_and_detect_speech command is also issued but with playing
>> silence.
>>
>> Obviously the new command will stop the utterance speaking of the
>> previous command if it is not finished.
>>
>> My question is
>>
>> will the new play_and_detect_speech command also stop the previous ASR
>> listening
>> or will there be many ASR listening channel and sending speech data (or
>> silence) to ASR server?
>>
>> Do I need to explicitly issue a "stop" commnad before issuing a new
>> play_and_detect_speech?
>> If yes, how to do that, by "detect_speech stop"?
>>
>> Recently there is a network traffice problem (lots of connections /data
>> transportation to the ASR server machine)
>> when I am running my application. I am not sure if this is because of
>> other issues
>> or because of my new changes to the way of using play_and_detect_speech.
>>
>> Please any one could shed a light on this?
>>
>> Many thanks!
>>
>> Xing
>>
>>
>> ------------------------------
>>
>>  *Heriot-Watt University is the Sunday Times Scottish University of the
>> Year 2011-2012.*
>>
>>  We invite research leaders and ambitious early career researchers to
>> join us in leading and driving research in key inter-disciplinary themes.
>> Please see www.hw.ac.uk/researchleaders<http://www.hw.ac.uk/researchleaders> for further information and how
>> to apply.
>>
>>  Heriot-Watt University is a Scottish charity registered under charity
>> number SC000278.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
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>> 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
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users<mailto: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<http://lists.freeswitch.org/mailman/options/freeswitch-usershttp:/www.freeswitch.org>
>
>
>
> ------------------------------
>
>  *Heriot-Watt University is the Sunday Times Scottish University of the
> Year 2011-2012.*
>
>  We invite research leaders and ambitious early career researchers to join
> us in leading and driving research in key inter-disciplinary themes. Please
> see www.hw.ac.uk/researchleaders<http://www.hw.ac.uk/researchleaders> for further information and how to
> apply.
>
>  Heriot-Watt University is a Scottish charity registered under charity
> number SC000278.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-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
>
>

________________________________

Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012.

We invite research leaders and ambitious early career researchers to join us in leading and driving research in key inter-disciplinary themes. Please see www.hw.ac.uk/researchleaders<http://www.hw.ac.uk/researchleaders> for further information and how to apply.

Heriot-Watt University is a Scottish charity registered under charity number SC000278.



-- 
Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012

We invite research leaders and ambitious early career researchers to 
join us in leading and driving research in key inter-disciplinary themes. 
Please see www.hw.ac.uk/researchleaders for further information and how
to apply.

Heriot-Watt University is a Scottish charity
registered under charity number SC000278.

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From cmrienzo at gmail.com  Wed Aug  1 23:06:42 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 1 Aug 2012 15:06:42 -0400
Subject: [Freeswitch-users] Am I using play_and_detect_speech correctly?
In-Reply-To: <D346F0E06A1D9341B7FF1FC1B525698C5C1299@ex10.mail.win.hw.ac.uk>
References: <D346F0E06A1D9341B7FF1FC1B525698C5C128A@ex10.mail.win.hw.ac.uk>
	<CAD-qRkhF-=6y6xWK0eyuZN3FSCUw7qAhHwyHbnJnXTmoZxZXrA@mail.gmail.com>
	<50179069.6020700@hw.ac.uk>
	<CAD-qRkiL-KFF-tkKk3Ni=eOn+A6dp5eKxe61PK=6_m5N_WBaFw@mail.gmail.com>
	<D346F0E06A1D9341B7FF1FC1B525698C5C1291@ex10.mail.win.hw.ac.uk>
	<6A6B4C284AD15042B429EB9D904544AD02304F6260@NY1-EXMB-01.ip-soft.net>
	<D346F0E06A1D9341B7FF1FC1B525698C5C1299@ex10.mail.win.hw.ac.uk>
Message-ID: <CAD-qRkjr=yT8dR6umXPOZ3-io3SDtNggZjW-mvfPSFjxmn+PnQ@mail.gmail.com>

The basic procedure for barge in is:

1. detect_speech unimrcp
{start-input-timers=false,no-input-timeout=5000,recognition-timeout=5000}builtin:grammar/boolean?language=en-US;y=1;n=2
2. playback say:please say yes or no. please say no or yes. please say
something!
3. handle begin-speaking event
4. break
5. when playback finishes...  detect_speech start_input_timers
6. handle detected-speech event

This is pretty much what play_and_detect_speech already does... see
switch_ivr_play_and_detect_speech() in switch_ivr_async.c if you know C.

Chris


On Wed, Aug 1, 2012 at 1:49 PM, Liu, Xingkun <X.Liu at hw.ac.uk> wrote:

> **
>
> Thanks for your response, Hector!
>
> Yeah, I am using detect_speech via a similar way to yours.
>
> What I am more interested in is to use detect_speech app to handle user's
> barge-in.
>
> After Chris mentioned the barge-in can be also handled by detect_speech I
> gave it a further thinking.
> Yeah, I could first "speak" the utterance and immediately resume ASR, then
> try to catch the begin_speaking event,
> then stop the TTS -- using this way to handle the user barge-in.
> (Chris, you may have a better idea, would you please let me know if you
> do?)
>
> One thing I am worry about is that stopping currently playing media or
> utterance seems not work for me.
> When I recently try "api uuid_break <uuid>" it stopped currently playing
> music but also stopped playing following utterances
> which I sent to TTS soon later on after uuid_break.
>
> Anyway I will try it further again and let you all know what I will get.
>
> Cheers,
>
> Xing
>
>
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From mario_fs at mgtech.com  Thu Aug  2 00:07:42 2012
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 1 Aug 2012 13:07:42 -0700
Subject: [Freeswitch-users] Please help - How to use different variables in
	different bridge groups
Message-ID: <4DF6B13C-5664-4C76-9B7A-A6CDA0FB7270@mgtech.com>

What I need: to have different variable/settings for each target group in a bridge command, and allow leg_delay_start to work. It's related to trying to fix other issues and holding up progress for them. Thanks for any help,
Mario G

I scoured the wiki and tried many things. Normally, enterprise syntax would solve this, but there is a catch: you can't use leg_delay_start which I need, it is ignored when using enterprise. I found that using brackets [] only applies the variable to the first user in the group, not the whole group so that won't work.

What I have now but need to have alert_info moved/apply to the mgt group only: 
<action application="bridge" data="{originate_timeout=45,alert_info=n=${lua_ringtone}}${group_call(mgtbria@${domain_name}+A)},${group_call(mgt@${domain_name}+A)},[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551212,[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551313"/>
    

 This solves variables for groups but breaks leg_delay_start, etc.:
<action application="bridge" data="<originate_timeout=45>}${group_call(mgtbria@${domain_name}+A)}:_:{alert_info=n=${lua_ringtone}${group_call(mgt@${domain_name}+A)}:_:[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551212,[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551313"/>
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From cogs66 at gmail.com  Thu Aug  2 02:44:12 2012
From: cogs66 at gmail.com (cogs66)
Date: Wed, 1 Aug 2012 15:44:12 -0700 (PDT)
Subject: [Freeswitch-users] No call_uuid in ringing state unix odbc for
	PostgreSQL
In-Reply-To: <CAMwZ-t+3-6hSzzgYPivCE_bpboWWQwgnPCXg9nZqC6H2CY5spQ@mail.gmail.com>
References: <1343558012157-7581267.post@n2.nabble.com>
	<CAMwZ-t+3-6hSzzgYPivCE_bpboWWQwgnPCXg9nZqC6H2CY5spQ@mail.gmail.com>
Message-ID: <1343861052930-7581425.post@n2.nabble.com>

Thanks Gabe

I am not sure where the issue lies to be honest. It is odd why it works on
all my other installs.

Andy



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/No-call-uuid-in-ringing-state-unix-odbc-for-PostgreSQL-tp7581267p7581425.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From govoiper at gmail.com  Thu Aug  2 08:49:02 2012
From: govoiper at gmail.com (SamyGo)
Date: Thu, 2 Aug 2012 09:49:02 +0500
Subject: [Freeswitch-users] Nibble Billing deducted balance from account
 even call doesn't come to my mobile number
In-Reply-To: <50194496.10804@bestnet.kharkov.ua>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<50194496.10804@bestnet.kharkov.ua>
Message-ID: <CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>

You might wanna change your dialplan to something like this:

<action application="bridge" data="{
ignore_early_media=true }sofia/gateway/${default_gateway}/${dialed_number}"/>
That's How I;ve done it, no wonder I use LUA but the dial-string is exactly
as below.

I think there is a difference if you do it like above, It will set the
ignore_early_media on the B-leg session and  not on A-leg.

This a snippet from the FS-1.6 book: *page:186*

"Curly brackets are used "globally" for the duration of a call. Take the
following example where we are bridging a call to Darren's cell phone,
203-829-3150. We
only want to ring the phone for 20 seconds, to avoid hitting voicemail.

<application action="bridge"
data="{call_timeout=20}sofia/my_provider/2038293150">

*The variable in brackets is utilized on the newly setup channel,
sofia/my_provider/2038293150.*"


Regards,
Sammy


On Wed, Aug 1, 2012 at 8:00 PM, Evgeniy Movlyan
<evgeniy at bestnet.kharkov.ua>wrote:

> It did not work for me, i got the same message.
> My outbond extension:
>
> <extension name="AUTH_rad">
>   <condition field="${toll_allow}" expression="international"/>
>    <condition field="${AUTH_RESULT_RADIUS}" expression="^NOK$"
> break="on-true">
>     <action application="lua" data="callfailed.lua ${caller_id_number}"/>
>    </condition>
>    <condition field="${AUTH_RESULT_RADIUS}" expression="^OK$">
>     <action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>     <action application="set"
> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>     <action application="set" data="ignore_early_media=true"/>
>     <action application="bridge"
> data="sofia/gateway/${default_gateway}/${dialed_number}"/>
>    </condition>
> </extension>
>
>
> 31.07.2012 11:06, SamyGo ?????:
> > Hi,
> > Well its works perfect for me, do you guys have ignore_early_media set in
> > your outbound string, if no then set it and then see what happens.
> > On Jul 31, 2012 12:50 PM, "Evgeniy Movlyan"<evgeniy at bestnet.kharkov.ua>
> > wrote:
> >
> >> I have the same problem. When i am calling from one my extension to
> >> another all is ok, but when i am calling to external number i got this
> >> message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is not in
> >> answered state".
> >>
> >> 31.07.2012 10:35, virendra bhati ?????:
> >>> mod_nibblebill.c:465 Not billing
> >>> 97183008 - call is not in answered state
> >>
> >> --
> >> Evgeniy Movlyan,
> >> BestNet Ltd.
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Evgeniy Movlyan,
> BestNet Ltd.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From andrew at cassidywebservices.co.uk  Thu Aug  2 12:36:16 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 2 Aug 2012 09:36:16 +0100
Subject: [Freeswitch-users] FreeSWITCH dying with no core file
In-Reply-To: <CAE2A9TCosukvTHHa6YWWYskmNwFH8fheBQjceGPWpPy6r_mf_Q@mail.gmail.com>
References: <CAE2A9TCfrCw1=9ScVzsCcnP0CvaYVpMbHPAm19ChwY3rEQrN8A@mail.gmail.com>
	<5013012C.9050003@freeswitch.org>
	<CAE2A9TDGczBNAss75AE5pVqMutotrkec7uvPWmJwjkV+t8+AZA@mail.gmail.com>
	<CAMbW9oKSFLCF-32rOD1g2wA4C_0s2DL-mSd8JmqWRbyTA+ed=Q@mail.gmail.com>
	<CAE2A9TCosukvTHHa6YWWYskmNwFH8fheBQjceGPWpPy6r_mf_Q@mail.gmail.com>
Message-ID: <CAMbW9oLtigMhRdJdfOCYBXhrK0-rrs-TMq2vC=bHERdpU+1UCg@mail.gmail.com>

Thanks. Ok it doesn't sound related to my problem unfortunately :( I'll
have to persevere!

On 1 August 2012 17:04, Adam Johnston <ajohnston at blimessaging.com> wrote:

> Andrew,
>
> I loaded the core file into gdb and looked at the backtrace. It turns out
> that the problem was not in the FreeSWITCH code but rather in LuaSocket,
> and the error was thrown while I was caching fax results in a key-value
> store using a Lua script. My issue was very similar to this Jira:
> http://jira.freeswitch.org/browse/FS-2893. I recompiled LuaSocket with
> the FreeSWITCH includes and am hoping that fixes things.
>
> Adam
>
>
> On Wed, Aug 1, 2012 at 11:27 AM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
>> Hi, what have you done with the core file?
>>
>> On 31 July 2012 21:16, Adam Johnston <ajohnston at blimessaging.com> wrote:
>>
>>> That did it. My sporadic crash happened again and, sure enough, there's
>>> a core file.
>>>
>>> Thanks again,
>>> Adam Johnston
>>>
>>>
>>> On Fri, Jul 27, 2012 at 4:59 PM, Stefan Knoblich <stkn at freeswitch.org>wrote:
>>>
>>>> -----BEGIN PGP SIGNED MESSAGE-----
>>>> Hash: SHA1
>>>>
>>>> On 07/27/12 21:34, Adam Johnston wrote:
>>>> > I use FreeSWITCH exclusively for high-volume fax, both sending and
>>>> receiving, and I've been running into a sporadic issue where the FreeSWITCH
>>>> process will die with nothing in the logs to
>>>> > indicate an issue and no core file anywhere on the system. I'm
>>>> setting "ulimit -c unlimited" in my init script, so I would expect to see a
>>>> core file. Version: FreeSWITCH Version
>>>> > 1.2.0-rc2+git~20120713T162602Z~0afd7318bd+unclean~20120713T182401Z
>>>>
>>>> You need to set /proc/sys/fs/suid_dumpable to a non-zero value (see man
>>>> 5 proc), to enable core dumping for processes
>>>> that switch their UID on startup (such as freeswitch with -u option).
>>>>
>>>> By default, the freeswitch process will try to put the core file into
>>>> the CWD (current work directory),
>>>> so make sure the UID it is running as can write there (or "cd" into a
>>>> different directory before starting it).
>>>> You might also want to check out man 5 core, for all the other options
>>>> (e.g. piping core dumps into a custom handler, with core_pattern).
>>>>
>>>> -----BEGIN PGP SIGNATURE-----
>>>> Version: GnuPG v2.0.19 (GNU/Linux)
>>>> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
>>>>
>>>> iEYEARECAAYFAlATASwACgkQjiIIAK4rYUrjvQCgh1grtcBfPVhbZNw8tdQpsyda
>>>> 614Amwb1pKDFL93dU33cy2wj1FNYYnzJ
>>>> =0Xqk
>>>> -----END PGP SIGNATURE-----
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>>
>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>  *03300 100 961
>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From odermann at googlemail.com  Thu Aug  2 13:26:22 2012
From: odermann at googlemail.com (Dennis)
Date: Thu, 2 Aug 2012 11:26:22 +0200
Subject: [Freeswitch-users] Fax (RX): Missing name/header variable in
	socket!?
Message-ID: <CALaOL+hYAxkVTEOLBCpCZB5H6LbZaLH_hW-UiagvP_XYAKOQbQ@mail.gmail.com>

hi,

we are receiving many faxes (RX), but we are missing a variable in the socket.

we do get the "remote station id", which is the fax-number itself, but
we need the name/header of the incoming fax (for example: COMPANYNAME
INC.).


thanks for your help
dennis


From evgeniy at bestnet.kharkov.ua  Thu Aug  2 14:48:20 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Thu, 02 Aug 2012 13:48:20 +0300
Subject: [Freeswitch-users] Nibble Billing deducted balance from account
 even call doesn't come to my mobile number
In-Reply-To: <CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>	<50178DB7.7000606@bestnet.kharkov.ua>	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>	<50194496.10804@bestnet.kharkov.ua>
	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>
Message-ID: <501A5AF4.9020008@bestnet.kharkov.ua>

I changed my dialplan, but got the same message =(

02.08.2012 07:49, SamyGo ?????:
> You might wanna change your dialplan to something like this:
>
> <action application="bridge" data="{
> ignore_early_media=true }sofia/gateway/${default_gateway}/${dialed_number}"/>
> That's How I;ve done it, no wonder I use LUA but the dial-string is exactly
> as below.
>
> I think there is a difference if you do it like above, It will set the
> ignore_early_media on the B-leg session and  not on A-leg.
>
> This a snippet from the FS-1.6 book: *page:186*
>
> "Curly brackets are used "globally" for the duration of a call. Take the
> following example where we are bridging a call to Darren's cell phone,
> 203-829-3150. We
> only want to ring the phone for 20 seconds, to avoid hitting voicemail.
>
> <application action="bridge"
> data="{call_timeout=20}sofia/my_provider/2038293150">
>
> *The variable in brackets is utilized on the newly setup channel,
> sofia/my_provider/2038293150.*"
>
>
> Regards,
> Sammy
>
>
> On Wed, Aug 1, 2012 at 8:00 PM, Evgeniy Movlyan
> <evgeniy at bestnet.kharkov.ua>wrote:
>
>> It did not work for me, i got the same message.
>> My outbond extension:
>>
>> <extension name="AUTH_rad">
>>    <condition field="${toll_allow}" expression="international"/>
>>     <condition field="${AUTH_RESULT_RADIUS}" expression="^NOK$"
>> break="on-true">
>>      <action application="lua" data="callfailed.lua ${caller_id_number}"/>
>>     </condition>
>>     <condition field="${AUTH_RESULT_RADIUS}" expression="^OK$">
>>      <action application="set"
>> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>>      <action application="set"
>> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>>      <action application="set" data="ignore_early_media=true"/>
>>      <action application="bridge"
>> data="sofia/gateway/${default_gateway}/${dialed_number}"/>
>>     </condition>
>> </extension>
>>
>>
>> 31.07.2012 11:06, SamyGo ?????:
>>> Hi,
>>> Well its works perfect for me, do you guys have ignore_early_media set in
>>> your outbound string, if no then set it and then see what happens.
>>> On Jul 31, 2012 12:50 PM, "Evgeniy Movlyan"<evgeniy at bestnet.kharkov.ua>
>>> wrote:
>>>
>>>> I have the same problem. When i am calling from one my extension to
>>>> another all is ok, but when i am calling to external number i got this
>>>> message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is not in
>>>> answered state".
>>>>
>>>> 31.07.2012 10:35, virendra bhati ?????:
>>>>> mod_nibblebill.c:465 Not billing
>>>>> 97183008 - call is not in answered state
>>>>
>>>> --
>>>> Evgeniy Movlyan,
>>>> BestNet Ltd.
>>>>
>>>>
>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> --
>> Evgeniy Movlyan,
>> BestNet Ltd.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Evgeniy Movlyan,
BestNet Ltd.


From govoiper at gmail.com  Thu Aug  2 14:58:07 2012
From: govoiper at gmail.com (SamyGo)
Date: Thu, 2 Aug 2012 15:58:07 +0500
Subject: [Freeswitch-users] Nibble Billing deducted balance from account
 even call doesn't come to my mobile number
In-Reply-To: <501A5AF4.9020008@bestnet.kharkov.ua>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<50194496.10804@bestnet.kharkov.ua>
	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>
	<501A5AF4.9020008@bestnet.kharkov.ua>
Message-ID: <CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>

Ok,
Paste the whole call console log and share via pastebin.
Maybe find some other thing breaking it.
BR
Sammy
On Aug 2, 2012 3:52 PM, "Evgeniy Movlyan" <evgeniy at bestnet.kharkov.ua>
wrote:

> I changed my dialplan, but got the same message =(
>
> 02.08.2012 07:49, SamyGo ?????:
> > You might wanna change your dialplan to something like this:
> >
> > <action application="bridge" data="{
> > ignore_early_media=true
> }sofia/gateway/${default_gateway}/${dialed_number}"/>
> > That's How I;ve done it, no wonder I use LUA but the dial-string is
> exactly
> > as below.
> >
> > I think there is a difference if you do it like above, It will set the
> > ignore_early_media on the B-leg session and  not on A-leg.
> >
> > This a snippet from the FS-1.6 book: *page:186*
> >
> > "Curly brackets are used "globally" for the duration of a call. Take the
> > following example where we are bridging a call to Darren's cell phone,
> > 203-829-3150. We
> > only want to ring the phone for 20 seconds, to avoid hitting voicemail.
> >
> > <application action="bridge"
> > data="{call_timeout=20}sofia/my_provider/2038293150">
> >
> > *The variable in brackets is utilized on the newly setup channel,
> > sofia/my_provider/2038293150.*"
> >
> >
> > Regards,
> > Sammy
> >
> >
> > On Wed, Aug 1, 2012 at 8:00 PM, Evgeniy Movlyan
> > <evgeniy at bestnet.kharkov.ua>wrote:
> >
> >> It did not work for me, i got the same message.
> >> My outbond extension:
> >>
> >> <extension name="AUTH_rad">
> >>    <condition field="${toll_allow}" expression="international"/>
> >>     <condition field="${AUTH_RESULT_RADIUS}" expression="^NOK$"
> >> break="on-true">
> >>      <action application="lua" data="callfailed.lua
> ${caller_id_number}"/>
> >>     </condition>
> >>     <condition field="${AUTH_RESULT_RADIUS}" expression="^OK$">
> >>      <action application="set"
> >> data="effective_caller_id_number=${outbound_caller_id_number}"/>
> >>      <action application="set"
> >> data="effective_caller_id_name=${outbound_caller_id_name}"/>
> >>      <action application="set" data="ignore_early_media=true"/>
> >>      <action application="bridge"
> >> data="sofia/gateway/${default_gateway}/${dialed_number}"/>
> >>     </condition>
> >> </extension>
> >>
> >>
> >> 31.07.2012 11:06, SamyGo ?????:
> >>> Hi,
> >>> Well its works perfect for me, do you guys have ignore_early_media set
> in
> >>> your outbound string, if no then set it and then see what happens.
> >>> On Jul 31, 2012 12:50 PM, "Evgeniy Movlyan"<evgeniy at bestnet.kharkov.ua
> >
> >>> wrote:
> >>>
> >>>> I have the same problem. When i am calling from one my extension to
> >>>> another all is ok, but when i am calling to external number i got this
> >>>> message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is not in
> >>>> answered state".
> >>>>
> >>>> 31.07.2012 10:35, virendra bhati ?????:
> >>>>> mod_nibblebill.c:465 Not billing
> >>>>> 97183008 - call is not in answered state
> >>>>
> >>>> --
> >>>> Evgeniy Movlyan,
> >>>> BestNet Ltd.
> >>>>
> >>>>
> >>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> Join Us At ClueCon - Aug 7-9, 2012
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> Join Us At ClueCon - Aug 7-9, 2012
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >> --
> >> Evgeniy Movlyan,
> >> BestNet Ltd.
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Evgeniy Movlyan,
> BestNet Ltd.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From evgeniy at bestnet.kharkov.ua  Thu Aug  2 15:12:05 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Thu, 02 Aug 2012 14:12:05 +0300
Subject: [Freeswitch-users] Nibble Billing deducted balance from account
 even call doesn't come to my mobile number
In-Reply-To: <CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>	<50178DB7.7000606@bestnet.kharkov.ua>	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>	<50194496.10804@bestnet.kharkov.ua>	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>	<501A5AF4.9020008@bestnet.kharkov.ua>
	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>
Message-ID: <501A6085.2020308@bestnet.kharkov.ua>

http://pastebin.com/enEmLVJz

02.08.2012 13:58, SamyGo ?????:
> Ok,
> Paste the whole call console log and share via pastebin.
> Maybe find some other thing breaking it.
> BR
> Sammy
> On Aug 2, 2012 3:52 PM, "Evgeniy Movlyan"<evgeniy at bestnet.kharkov.ua>
> wrote:
>
>> I changed my dialplan, but got the same message =(
>>
>> 02.08.2012 07:49, SamyGo ?????:
>>> You might wanna change your dialplan to something like this:
>>>
>>> <action application="bridge" data="{
>>> ignore_early_media=true
>> }sofia/gateway/${default_gateway}/${dialed_number}"/>
>>> That's How I;ve done it, no wonder I use LUA but the dial-string is
>> exactly
>>> as below.
>>>
>>> I think there is a difference if you do it like above, It will set the
>>> ignore_early_media on the B-leg session and  not on A-leg.
>>>
>>> This a snippet from the FS-1.6 book: *page:186*
>>>
>>> "Curly brackets are used "globally" for the duration of a call. Take the
>>> following example where we are bridging a call to Darren's cell phone,
>>> 203-829-3150. We
>>> only want to ring the phone for 20 seconds, to avoid hitting voicemail.
>>>
>>> <application action="bridge"
>>> data="{call_timeout=20}sofia/my_provider/2038293150">
>>>
>>> *The variable in brackets is utilized on the newly setup channel,
>>> sofia/my_provider/2038293150.*"
>>>
>>>
>>> Regards,
>>> Sammy
>>>
>>>
>>> On Wed, Aug 1, 2012 at 8:00 PM, Evgeniy Movlyan
>>> <evgeniy at bestnet.kharkov.ua>wrote:
>>>
>>>> It did not work for me, i got the same message.
>>>> My outbond extension:
>>>>
>>>> <extension name="AUTH_rad">
>>>>     <condition field="${toll_allow}" expression="international"/>
>>>>      <condition field="${AUTH_RESULT_RADIUS}" expression="^NOK$"
>>>> break="on-true">
>>>>       <action application="lua" data="callfailed.lua
>> ${caller_id_number}"/>
>>>>      </condition>
>>>>      <condition field="${AUTH_RESULT_RADIUS}" expression="^OK$">
>>>>       <action application="set"
>>>> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>>>>       <action application="set"
>>>> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>>>>       <action application="set" data="ignore_early_media=true"/>
>>>>       <action application="bridge"
>>>> data="sofia/gateway/${default_gateway}/${dialed_number}"/>
>>>>      </condition>
>>>> </extension>
>>>>
>>>>
>>>> 31.07.2012 11:06, SamyGo ?????:
>>>>> Hi,
>>>>> Well its works perfect for me, do you guys have ignore_early_media set
>> in
>>>>> your outbound string, if no then set it and then see what happens.
>>>>> On Jul 31, 2012 12:50 PM, "Evgeniy Movlyan"<evgeniy at bestnet.kharkov.ua
>>>
>>>>> wrote:
>>>>>
>>>>>> I have the same problem. When i am calling from one my extension to
>>>>>> another all is ok, but when i am calling to external number i got this
>>>>>> message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is not in
>>>>>> answered state".
>>>>>>
>>>>>> 31.07.2012 10:35, virendra bhati ?????:
>>>>>>> mod_nibblebill.c:465 Not billing
>>>>>>> 97183008 - call is not in answered state
>>>>>>
>>>>>> --
>>>>>> Evgeniy Movlyan,
>>>>>> BestNet Ltd.
>>>>>>
>>>>>>
>>>>
>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>> --
>>>> Evgeniy Movlyan,
>>>> BestNet Ltd.
>>>>
>>>>
>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> --
>> Evgeniy Movlyan,
>> BestNet Ltd.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Evgeniy Movlyan,
BestNet Ltd.


From lloyd.aloysius at gmail.com  Thu Aug  2 15:34:22 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Thu, 2 Aug 2012 07:34:22 -0400
Subject: [Freeswitch-users] ESL - phpmod compile Error
In-Reply-To: <CAKzWOxVYu2HYAJ2X=jAdbJW=dZmZAniNBxtmNhq6057FuXCdBQ@mail.gmail.com>
References: <CAGvuWYa7c7Fy+B8gKNU4cSFLT6HtjYpC_03UvqezRRJSw4dXhw@mail.gmail.com>
	<CAKzWOxVYu2HYAJ2X=jAdbJW=dZmZAniNBxtmNhq6057FuXCdBQ@mail.gmail.com>
Message-ID: <CAGvuWYZS7702-XZC0Y_oEqqem2atnQDk5z+N-+81eq889WYK5g@mail.gmail.com>

Michael,

I already installed the   libedit-dev . Same error

Thanks
Lloyd



On Wed, Aug 1, 2012 at 12:18 PM, Michael Collins <msc at freeswitch.org> wrote:

> I suspect this line is the key piece of information:
> /usr/bin/ld: cannot find -ledit
>
> If I read that correctly you need to install libedit-dev. Give that a try
> and let us know.
>
> -MC
>
> On Tue, Jul 31, 2012 at 11:20 PM, Lloyd Aloysius <lloyd.aloysius at gmail.com
> > wrote:
>
>> Hi All:
>>
>> when I compile the ESL module for php . OS - Centos 6.2 / 64 bit
>>
>> libs/esl
>>
>> make phpmod
>>
>> I got the following Errors.
>>
>>   make phpmod
>> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
>> CFLAGS="-I/usr/src/freeswitch.git/libs/esl/src/include -DHAVE_EDITLINE -g
>> -ggdb -I../../libs/libedit/src/ -fPIC -O2"
>> CXXFLAGS="-I/usr/src/freeswitch.git/libs/esl/src/include -DHAVE_EDITLINE -g
>> -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php
>> make[1]: Entering directory `/usr/src/freeswitch.git/libs/esl/php'
>> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib6464 -lcrypt
>> -lcrypt -ledit -lncurses -lm -ldl -lnsl -lm -ldl -ldl -lm -lcrypt -lm
>> -lcrypt -lpthread -o ESL.so -L.
>> /usr/bin/ld: cannot find -ledit
>> collect2: ld returned 1 exit status
>> make[1]: *** [ESL.so] Error 1
>> make[1]: Leaving directory `/usr/src/freeswitch.git/libs/esl/php'
>> make: *** [phpmod] Error 2
>>
>>
>> How to solve this issue.
>>
>> Thanks
>> Lloyd
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nathandownes at hotmail.com  Thu Aug  2 16:13:40 2012
From: nathandownes at hotmail.com (Mr Nathan Downes)
Date: Thu, 2 Aug 2012 22:13:40 +1000
Subject: [Freeswitch-users] multi-tenant setup - call goes to wrong place
In-Reply-To: <0d7301cd6fe6$309450c0$91bcf240$@hotmail.com>
References: <03e401cd6be9$52f19b10$f8d4d130$@hotmail.com>
	<0d7301cd6fe6$309450c0$91bcf240$@hotmail.com>
Message-ID: <BLU170-DS2706D7B13702D8EA4BE118D7CB0@phx.gbl>

I had setup shared line appearance on one tenant, so had to re-enable it for
them to make calls, further investigation shows that manage-presence causes
the problem and it only affects domain listed in presence-hosts.  I took
domain3.com out of presence-hosts and when I call from 101 at domain3.com to
201 I do not get 201 at domain2.com

 

If 300 at domain1.com calls 201 they get sent to 201 at domain2.com

 

Is this a bug?  

 

I tried updated to latest head but there seems to be an issue with that that
even when people are registered as per normal, calling them internally or
routing to them from a did reports user_not_registered and they just get
voicemail, doesn't affect outbound though..  

 

I did have time to confirm the issue is still in latest head though

 

From: Mr Nathan Downes [mailto:nathandownes at hotmail.com] 
Sent: Wednesday, 1 August 2012 11:04 PM
To: 'FreeSWITCH Users Help'
Subject: FW: multi-tenant setup - call goes to wrong place

 

Hi,

 

I ended up locating the problem.. it appears when I was playing with
shared_presence.. to try and get BLF working for certain tenants. What I
thought would help from googling caused this issue somehow, below is what I
enabled to cause the problem.

 

In internal.xml

 

<!--enable to use presence -->

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

    <!-- send a presence probe on each register to query devices to send
presence instead of sending presence with less info -->

    <!--<param name="presence-probe-on-register" value="true"/>-->
this one enabled

    <!--<param name="manage-shared-appearance" value="false"/>-->
this one true + enabled

    <!-- used to share presence info across sofia profiles -->

    <!-- Name of the db to use for this profile -->

    <!-- param name="dbname" value="shared_presence"/> -->
this one  + enabled

    <param name="presence-hosts"
value="domain1.com,domain2.com,domain3.com,$${local_ip_v4}"/>

 

Is this actually required to use BLF for a certain domain?

 

 

From: Mr Nathan Downes [mailto:nathandownes at hotmail.com] 
Sent: Friday, 27 July 2012 9:17 PM
To: 'FreeSWITCH Users Help'
Subject: multi-tenant setup - call goes to wrong place

 

Hi,

 

I have a multi tenant setup with fusionpbx, for one domain I use extensions
that don't register just for voicemail i.e 201 at domain1.com , when someone
from xx at domain1.com calls 201 everything appears to go correctly, domain is
set and call goes to 201 at domain1.com but when it first becomes an ip it
sends to 201 at domain2.ip.address.  It doesn't seem to happen with all
extensions numbered the same in different domains..

 

Any idea what to look for??

 

Log as follows.

 

2012-07-26 09:37:35.178096 [NOTICE] switch_channel.c:926 New Channel
sofia/internal/306 at domain1.com [b62db898-d6b1-11e1-be15-73f389528c7f]

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/306 at domain1.com) Running State Change CS_NEW

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:403
(sofia/internal/306 at domain1.com) State NEW

2012-07-26 09:37:35.178096 [DEBUG] sofia.c:5838 Channel
sofia/internal/306 at domain1.com entering state [received][100]

2012-07-26 09:37:35.178096 [DEBUG] sofia.c:5849 Remote SDP:

v=0

o=- 60255882 60255882 IN IP4 115.64.93.203

s=-

c=IN IP4 115.64.93.203

t=0 0

m=audio 55938 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

 

2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash

2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3919 Deciding whether to
pass zrtp-hash between legs

2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ
not set, so not propagating zrtp-hash

2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:5034 Audio Codec Compare
[PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]

2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3020 Set Codec
sofia/internal/306 at domain1.com PCMA/8000 20 ms 160 samples 64000 bits

2012-07-26 09:37:35.178096 [DEBUG] switch_core_codec.c:111
sofia/internal/306 at domain1.com Original read codec set to PCMA:8

2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:5155 Set 2833 dtmf send/recv
payload to 101

2012-07-26 09:37:35.178096 [DEBUG] sofia.c:6077
(sofia/internal/306 at domain1.com) State Change CS_NEW -> CS_INIT

2012-07-26 09:37:35.178096 [DEBUG] switch_core_session.c:1228 Send signal
sofia/internal/306 at domain1.com [BREAK]

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/306 at domain1.com) Running State Change CS_INIT

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/306 at domain1.com) State INIT

2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:85
sofia/internal/306 at domain1.com SOFIA INIT

2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:125
(sofia/internal/306 at domain1.com) State Change CS_INIT -> CS_ROUTING

2012-07-26 09:37:35.178096 [DEBUG] switch_core_session.c:1228 Send signal
sofia/internal/306 at domain1.com [BREAK]

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/306 at domain1.com) State INIT going to sleep

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/306 at domain1.com) Running State Change CS_ROUTING

2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1919
(sofia/internal/306 at domain1.com) Callstate Change DOWN -> RINGING

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:433
(sofia/internal/306 at domain1.com) State ROUTING

2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:148
sofia/internal/306 at domain1.com SOFIA ROUTING

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:104
sofia/internal/306 at domain1.com Standard ROUTING

2012-07-26 09:37:35.178096 [INFO] mod_dialplan_xml.c:485 Processing
0286249343 <306>->201 in context domain1.com

Dialplan: sofia/internal/306 at domain1.com parsing [domain1.com->unloop]
continue=false

Dialplan: sofia/internal/306 at domain1.com parsing
[domain1.com->Local_Extension] continue=false

Dialplan: sofia/internal/306 at domain1.com Regex (PASS) [Local_Extension]
destination_number(201) =~ /(^\d{2,7}$)/ break=on-false

Dialplan: sofia/internal/306 at domain1.com Action set(dialed_extension=201) 

Dialplan: sofia/internal/306 at domain1.com Action export(dialed_extension=201)


Dialplan: sofia/internal/306 at domain1.com Action limit(hash ${domain_name}
201 ${limit_max} ${limit_destination}) 

Dialplan: sofia/internal/306 at domain1.com Action bind_meta_app(1 b s
execute_extension::dx XML features) 

Dialplan: sofia/internal/306 at domain1.com Action bind_meta_app(2 b s
record_session::/usr/local/freeswitch/recordings/archive/${strftime(%Y)}/${s
trftime(%b)}/${strftime(%d)}/${uuid}.wav) 

Dialplan: sofia/internal/306 at domain1.com Action bind_meta_app(3 b s
execute_extension::cf XML features) 

Dialplan: sofia/internal/306 at domain1.com Action bind_meta_app(4 b s
execute_extension::att_xfer XML features) 

Dialplan: sofia/internal/306 at domain1.com Action set(ringback=${us-ring}) 

Dialplan: sofia/internal/306 at domain1.com Action
set(transfer_ringback=local_stream://moh) 

Dialplan: sofia/internal/306 at domain1.com Action set(call_timeout=30) 

Dialplan: sofia/internal/306 at domain1.com Action
set(hangup_after_bridge=true) 

Dialplan: sofia/internal/306 at domain1.com Action set(continue_on_fail=true) 

Dialplan: sofia/internal/306 at domain1.com Action
hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe
r}) 

Dialplan: sofia/internal/306 at domain1.com Action
hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) 

Dialplan: sofia/internal/306 at domain1.com Action
set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}) 

Dialplan: sofia/internal/306 at domain1.com Action
hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) 

Dialplan: sofia/internal/306 at domain1.com Action
bridge(user/${user_data(${destination_number}@${domain_name} attr
id)}@${domain_name}) 

Dialplan: sofia/internal/306 at domain1.com Action answer() 

Dialplan: sofia/internal/306 at domain1.com Action sleep(1000) 

Dialplan: sofia/internal/306 at domain1.com Action voicemail(default
${domain_name} ${dialed_extension}) 

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:154
(sofia/internal/306 at domain1.com) State Change CS_ROUTING -> CS_EXECUTE

2012-07-26 09:37:35.178096 [DEBUG] switch_core_session.c:1228 Send signal
sofia/internal/306 at domain1.com [BREAK]

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:433
(sofia/internal/306 at domain1.com) State ROUTING going to sleep

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/306 at domain1.com) Running State Change CS_EXECUTE

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:440
(sofia/internal/306 at domain1.com) State EXECUTE

2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:241
sofia/internal/306 at domain1.com SOFIA EXECUTE

2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:196
sofia/internal/306 at domain1.com Standard EXECUTE

EXECUTE sofia/internal/306 at domain1.com set(call_direction=local)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [call_direction]=[local]

EXECUTE sofia/internal/306 at domain1.com set(open=true)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [open]=[true]

EXECUTE sofia/internal/306 at domain1.com
hash(insert/domain1.com-spymap/306/b62db898-d6b1-11e1-be15-73f389528c7f)

EXECUTE sofia/internal/306 at domain1.com
hash(insert/domain1.com-last_dial/306/201)

EXECUTE sofia/internal/306 at domain1.com
hash(insert/domain1.com-last_dial/global/b62db898-d6b1-11e1-be15-73f389528c7
f)

EXECUTE sofia/internal/306 at domain1.com set(RFC2822_DATE=Thu, 26 Jul 2012
09:37:35 +1000)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [RFC2822_DATE]=[Thu, 26 Jul 2012 09:37:35
+1000]

EXECUTE sofia/internal/306 at domain1.com set(dialed_extension=201)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [dialed_extension]=[201]

EXECUTE sofia/internal/306 at domain1.com export(dialed_extension=201)

2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1093 EXPORT
(export_vars) [dialed_extension]=[201]

EXECUTE sofia/internal/306 at domain1.com limit(hash domain1.com 201 2 !BUSY)

2012-07-26 09:37:35.178096 [INFO] switch_limit.c:126 incr called:
domain1.com_201 max:2, interval:0

2012-07-26 09:37:35.178096 [INFO] mod_hash.c:202 Usage for domain1.com_201
is now 1/2

EXECUTE sofia/internal/306 at domain1.com bind_meta_app(1 b s
execute_extension::dx XML features)

2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *1
execute_extension::dx XML features

EXECUTE sofia/internal/306 at domain1.com bind_meta_app(2 b s
record_session::/usr/local/freeswitch/recordings/archive/2012/Jul/26/b62db89
8-d6b1-11e1-be15-73f389528c7f.wav)

2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *2
record_session::/usr/local/freeswitch/recordings/archive/2012/Jul/26/b62db89
8-d6b1-11e1-be15-73f389528c7f.wav

EXECUTE sofia/internal/306 at domain1.com bind_meta_app(3 b s
execute_extension::cf XML features)

2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *3
execute_extension::cf XML features

EXECUTE sofia/internal/306 at domain1.com bind_meta_app(4 b s
execute_extension::att_xfer XML features)

2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *4
execute_extension::att_xfer XML features

EXECUTE sofia/internal/306 at domain1.com set(ringback=%(2000, 4000, 440.0,
480.0))

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [ringback]=[%(2000, 4000, 440.0, 480.0)]

EXECUTE sofia/internal/306 at domain1.com
set(transfer_ringback=local_stream://moh)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [transfer_ringback]=[local_stream://moh]

EXECUTE sofia/internal/306 at domain1.com set(call_timeout=30)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [call_timeout]=[30]

EXECUTE sofia/internal/306 at domain1.com set(hangup_after_bridge=true)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [hangup_after_bridge]=[true]

EXECUTE sofia/internal/306 at domain1.com set(continue_on_fail=true)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [continue_on_fail]=[true]

EXECUTE sofia/internal/306 at domain1.com
hash(insert/domain1.com-call_return/201/306)

EXECUTE sofia/internal/306 at domain1.com
hash(insert/domain1.com-last_dial_ext/201/b62db898-d6b1-11e1-be15-73f389528c
7f)

EXECUTE sofia/internal/306 at domain1.com set(called_party_callgroup=)

2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305
sofia/internal/306 at domain1.com SET [called_party_callgroup]=[UNDEF]

EXECUTE sofia/internal/306 at domain1.com
hash(insert/domain1.com-last_dial//b62db898-d6b1-11e1-be15-73f389528c7f)

EXECUTE sofia/internal/306 at domain1.com bridge(user/201 at domain1.com)

2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1047
sofia/internal/306 at domain1.com EXPORTING[export_vars]
[domain_name]=[domain1.com] to event

2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1047
sofia/internal/306 at domain1.com EXPORTING[export_vars]
[dialed_extension]=[201] to event

2012-07-26 09:37:35.178096 [DEBUG] switch_ivr_originate.c:1958 Parsing
global variables

2012-07-26 09:37:35.198097 [DEBUG] switch_channel.c:1047
sofia/internal/306 at domain1.com EXPORTING[export_vars]
[domain_name]=[domain1.com] to event

2012-07-26 09:37:35.198097 [DEBUG] switch_channel.c:1047
sofia/internal/306 at domain1.com EXPORTING[export_vars]
[dialed_extension]=[201] to event

2012-07-26 09:37:35.198097 [DEBUG] switch_ivr_originate.c:1958 Parsing
global variables

2012-07-26 09:37:35.198097 [DEBUG] switch_event.c:1470 Parsing variable
[sip_invite_domain]=[domain1.com]

2012-07-26 09:37:35.198097 [DEBUG] switch_event.c:1470 Parsing variable
[presence_id]=[201 at domain1.com]

2012-07-26 09:37:35.198097 [NOTICE] switch_channel.c:926 New Channel
sofia/internal/sip:201 at 110.143.31.101:1178
[b630218c-d6b1-11e1-be1d-73f389528c7f]

2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:4734
(sofia/internal/sip:201 at 110.143.31.101:1178) State Change CS_NEW -> CS_INIT

2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:1228 Send signal
sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK]

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/sip:201 at 110.143.31.101:1178) Running State Change CS_INIT

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/sip:201 at 110.143.31.101:1178) State INIT

2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:85
sofia/internal/sip:201 at 110.143.31.101:1178 SOFIA INIT

2012-07-26 09:37:35.198097 [DEBUG] sofia_glue.c:2602 Local SDP:

v=0

o=FreeSWITCH 1343234313 1343234314 IN IP4 203.174.163.226

s=FreeSWITCH

c=IN IP4 203.174.163.226

t=0 0

m=audio 25142 RTP/AVP 8 0 3 101 13

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:125
(sofia/internal/sip:201 at 110.143.31.101:1178) State Change CS_INIT ->
CS_ROUTING

2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:1228 Send signal
sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK]

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:424
(sofia/internal/sip:201 at 110.143.31.101:1178) State INIT going to sleep

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/sip:201 at 110.143.31.101:1178) Running State Change CS_ROUTING

2012-07-26 09:37:35.198097 [DEBUG] switch_channel.c:1919
(sofia/internal/sip:201 at 110.143.31.101:1178) Callstate Change DOWN ->
RINGING

2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:923 Send signal
sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK]

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:433
(sofia/internal/sip:201 at 110.143.31.101:1178) State ROUTING

2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:148
sofia/internal/sip:201 at 110.143.31.101:1178 SOFIA ROUTING

2012-07-26 09:37:35.198097 [DEBUG] switch_ivr_originate.c:67
(sofia/internal/sip:201 at 110.143.31.101:1178) State Change CS_ROUTING ->
CS_CONSUME_MEDIA

2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:1228 Send signal
sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK]

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:433
(sofia/internal/sip:201 at 110.143.31.101:1178) State ROUTING going to sleep

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:385
(sofia/internal/sip:201 at 110.143.31.101:1178) Running State Change
CS_CONSUME_MEDIA

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:452
(sofia/internal/sip:201 at 110.143.31.101:1178) State CONSUME_MEDIA

2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:452
(sofia/internal/sip:201 at 110.143.31.101:1178) State CONSUME_MEDIA going to
sleep

2012-07-26 09:37:35.198097 [DEBUG] sofia.c:5838 Channel
sofia/internal/sip:201 at 110.143.31.101:1178 entering state [calling][0]

2012-07-26 09:37:35.318096 [DEBUG] switch_core_session.c:923 Send signal
sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK]

2012-07-26 09:37:35.318096 [DEBUG] switch_core_session.c:923 Send signal
sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK]

2012-07-26 09:37:35.338097 [DEBUG] sofia.c:5838 Channel
sofia/internal/sip:201 at 110.143.31.101:1178 entering state [proceeding][180]

2012-07-26 09:37:35.338097 [NOTICE] sofia.c:5930 Ring-Ready
sofia/internal/sip:201 at 110.143.31.101:1178!

2012-07-26 09:37:35.338097 [INFO] switch_ivr_originate.c:1156 Sending early
media

2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3269 AUDIO RTP
[sofia/internal/306 at domain1.com] 203.174.163.226 port 26698 -> 115.64.93.203
port 55938 codec: 8 ms: 20

2012-07-26 09:37:35.338097 [DEBUG] switch_rtp.c:1680 Starting timer [soft]
160 bytes per 20ms

2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to
60ms (3 frames)

2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send
payload to 101

2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive
payload to 101

2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3566
sofia/internal/306 at domain1.com Set rtp dtmf delay to 40

2012-07-26 09:37:35.338097 [DEBUG] mod_sofia.c:2606 Ring SDP:

v=0

o=FreeSWITCH 1343232757 1343232758 IN IP4 203.174.163.226

s=FreeSWITCH

c=IN IP4 203.174.163.226

t=0 0

m=audio 26698 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

 

2012-07-26 09:37:35.338097 [NOTICE] mod_sofia.c:2609 Pre-Answer
sofia/internal/306 at domain1.com!

2012-07-26 09:37:35.338097 [DEBUG] switch_channel.c:3042
(sofia/internal/306 at domain1.com) Callstate Change RINGING -> EARLY

2012-07-26 09:37:35.338097 [DEBUG] switch_core_session.c:777 Send signal
sofia/internal/306 at domain1.com [BREAK]

 

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From david.villasmil.work at gmail.com  Thu Aug  2 19:43:33 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 2 Aug 2012 17:43:33 +0200
Subject: [Freeswitch-users] ESL Log to console
Message-ID: <CAFGRPVpu_3t1JG8SPqEb8J9JBn6VYOXL1Aa79u_7Hj-uO0WB0A@mail.gmail.com>

Hello Guys,

I'm starting off with ESL, which is cool, but I'm trying to log to the
console like Lua's "freeswitch.consoleLog("info", "BLAH BLAH");" doing it
like:

my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
$con->execute("log", "1, BlahBlah");


But nothing gets in the log files or console... and I can't find any
documentation as to how to log using "execute"...

any ideas?

Thanks!

David
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From msc at freeswitch.org  Thu Aug  2 20:14:38 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 2 Aug 2012 09:14:38 -0700
Subject: [Freeswitch-users] Fax (RX): Missing name/header variable in
	socket!?
In-Reply-To: <CALaOL+hYAxkVTEOLBCpCZB5H6LbZaLH_hW-UiagvP_XYAKOQbQ@mail.gmail.com>
References: <CALaOL+hYAxkVTEOLBCpCZB5H6LbZaLH_hW-UiagvP_XYAKOQbQ@mail.gmail.com>
Message-ID: <CAKzWOxX-Doz18y+otDTGkPOumUOnA0=tD85tdrY8QS+e1pToiQ@mail.gmail.com>

Can you pastebin a console debug log of this happening?
-MC

On Thu, Aug 2, 2012 at 2:26 AM, Dennis <odermann at googlemail.com> wrote:

> hi,
>
> we are receiving many faxes (RX), but we are missing a variable in the
> socket.
>
> we do get the "remote station id", which is the fax-number itself, but
> we need the name/header of the incoming fax (for example: COMPANYNAME
> INC.).
>
>
> thanks for your help
> dennis
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Aug  2 20:17:59 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 2 Aug 2012 09:17:59 -0700
Subject: [Freeswitch-users] ESL - phpmod compile Error
In-Reply-To: <CAGvuWYZS7702-XZC0Y_oEqqem2atnQDk5z+N-+81eq889WYK5g@mail.gmail.com>
References: <CAGvuWYa7c7Fy+B8gKNU4cSFLT6HtjYpC_03UvqezRRJSw4dXhw@mail.gmail.com>
	<CAKzWOxVYu2HYAJ2X=jAdbJW=dZmZAniNBxtmNhq6057FuXCdBQ@mail.gmail.com>
	<CAGvuWYZS7702-XZC0Y_oEqqem2atnQDk5z+N-+81eq889WYK5g@mail.gmail.com>
Message-ID: <CAKzWOxXOr8QKgPvy4tMir3jNR0q7Hzx1eyz_erZ0C6Ks6cYv6A@mail.gmail.com>

Well, google suggests you're not the first one with this issue. Raymond
made a suggestion to someone else in this thread:
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050006.html

Perhaps it's a different library name.

-MC

On Thu, Aug 2, 2012 at 4:34 AM, Lloyd Aloysius <lloyd.aloysius at gmail.com>wrote:

> Michael,
>
> I already installed the   libedit-dev . Same error
>
> Thanks
> Lloyd
>
>
>
> On Wed, Aug 1, 2012 at 12:18 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> I suspect this line is the key piece of information:
>> /usr/bin/ld: cannot find -ledit
>>
>> If I read that correctly you need to install libedit-dev. Give that a try
>> and let us know.
>>
>> -MC
>>
>> On Tue, Jul 31, 2012 at 11:20 PM, Lloyd Aloysius <
>> lloyd.aloysius at gmail.com> wrote:
>>
>>> Hi All:
>>>
>>> when I compile the ESL module for php . OS - Centos 6.2 / 64 bit
>>>
>>> libs/esl
>>>
>>> make phpmod
>>>
>>> I got the following Errors.
>>>
>>>   make phpmod
>>> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
>>> CFLAGS="-I/usr/src/freeswitch.git/libs/esl/src/include -DHAVE_EDITLINE -g
>>> -ggdb -I../../libs/libedit/src/ -fPIC -O2"
>>> CXXFLAGS="-I/usr/src/freeswitch.git/libs/esl/src/include -DHAVE_EDITLINE -g
>>> -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php
>>> make[1]: Entering directory `/usr/src/freeswitch.git/libs/esl/php'
>>> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib6464 -lcrypt
>>> -lcrypt -ledit -lncurses -lm -ldl -lnsl -lm -ldl -ldl -lm -lcrypt -lm
>>> -lcrypt -lpthread -o ESL.so -L.
>>> /usr/bin/ld: cannot find -ledit
>>> collect2: ld returned 1 exit status
>>> make[1]: *** [ESL.so] Error 1
>>> make[1]: Leaving directory `/usr/src/freeswitch.git/libs/esl/php'
>>> make: *** [phpmod] Error 2
>>>
>>>
>>> How to solve this issue.
>>>
>>> Thanks
>>> Lloyd
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Aug  2 20:25:05 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 2 Aug 2012 09:25:05 -0700
Subject: [Freeswitch-users] ESL Log to console
In-Reply-To: <CAFGRPVpu_3t1JG8SPqEb8J9JBn6VYOXL1Aa79u_7Hj-uO0WB0A@mail.gmail.com>
References: <CAFGRPVpu_3t1JG8SPqEb8J9JBn6VYOXL1Aa79u_7Hj-uO0WB0A@mail.gmail.com>
Message-ID: <CAKzWOxWT5J=HFJ8hB=bpo32ACtmmqrNYqNGSar=2incVFWej+w@mail.gmail.com>

David,

Do you have a call in progress at this point? If not then you'll need to
supply a uuid of a live call, as mentioned here:
http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute

Remember, "execute" means "execute dialplan application" so if there's no
channel then "execute" really doesn't mean a whole lot.

Alternatively you could try something like this:

$con->api("log","WARNING Don't cross the streams!!");

Remember this rule of thumb: pretty much anything you type at fs_cli is an
"API" and therefore you can use $con->api(), whereas anything that is a
diaplan application requires an actual channel on which to run
$con->execute().

Hope that makes sense... :)

-Michael

On Thu, Aug 2, 2012 at 8:43 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello Guys,
>
> I'm starting off with ESL, which is cool, but I'm trying to log to the
> console like Lua's "freeswitch.consoleLog("info", "BLAH BLAH");" doing it
> like:
>
> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
> $con->execute("log", "1, BlahBlah");
>
>
> But nothing gets in the log files or console... and I can't find any
> documentation as to how to log using "execute"...
>
> any ideas?
>
> Thanks!
>
> David
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From X.Liu at hw.ac.uk  Thu Aug  2 20:25:33 2012
From: X.Liu at hw.ac.uk (Liu, Xingkun)
Date: Thu, 2 Aug 2012 17:25:33 +0100
Subject: [Freeswitch-users] Am I using play_and_detect_speech correctly?
References: <D346F0E06A1D9341B7FF1FC1B525698C5C128A@ex10.mail.win.hw.ac.uk><CAD-qRkhF-=6y6xWK0eyuZN3FSCUw7qAhHwyHbnJnXTmoZxZXrA@mail.gmail.com><50179069.6020700@hw.ac.uk><CAD-qRkiL-KFF-tkKk3Ni=eOn+A6dp5eKxe61PK=6_m5N_WBaFw@mail.gmail.com><D346F0E06A1D9341B7FF1FC1B525698C5C1291@ex10.mail.win.hw.ac.uk><6A6B4C284AD15042B429EB9D904544AD02304F6260@NY1-EXMB-01.ip-soft.net><D346F0E06A1D9341B7FF1FC1B525698C5C1299@ex10.mail.win.hw.ac.uk>
	<CAD-qRkjr=yT8dR6umXPOZ3-io3SDtNggZjW-mvfPSFjxmn+PnQ@mail.gmail.com>
Message-ID: <D346F0E06A1D9341B7FF1FC1B525698C5C129B@ex10.mail.win.hw.ac.uk>

Many thanks, Chris!

I will have tries on it later on.

In my original question regarding to too many network connections/much traffic issues,
I re-setup the destination machine OS the problem seems to have been solved.

So the play_and_detect_speech APP look likes working fine, so I have not yet tried the workaround 
by issuing "detect_speech pause". I will try it later if there was something unusual happening 
to me after more tests.

So for now I think the problem has been resolved.

Thanks again!

Xing



-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Christopher Rienzo
Sent: Wed 8/1/2012 20:06
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Am I using play_and_detect_speech correctly?
 
The basic procedure for barge in is:

1. detect_speech unimrcp
{start-input-timers=false,no-input-timeout=5000,recognition-timeout=5000}builtin:grammar/boolean?language=en-US;y=1;n=2
2. playback say:please say yes or no. please say no or yes. please say
something!
3. handle begin-speaking event
4. break
5. when playback finishes...  detect_speech start_input_timers
6. handle detected-speech event

This is pretty much what play_and_detect_speech already does... see
switch_ivr_play_and_detect_speech() in switch_ivr_async.c if you know C.

Chris


On Wed, Aug 1, 2012 at 1:49 PM, Liu, Xingkun <X.Liu at hw.ac.uk> wrote:

> **
>
> Thanks for your response, Hector!
>
> Yeah, I am using detect_speech via a similar way to yours.
>
> What I am more interested in is to use detect_speech app to handle user's
> barge-in.
>
> After Chris mentioned the barge-in can be also handled by detect_speech I
> gave it a further thinking.
> Yeah, I could first "speak" the utterance and immediately resume ASR, then
> try to catch the begin_speaking event,
> then stop the TTS -- using this way to handle the user barge-in.
> (Chris, you may have a better idea, would you please let me know if you
> do?)
>
> One thing I am worry about is that stopping currently playing media or
> utterance seems not work for me.
> When I recently try "api uuid_break <uuid>" it stopped currently playing
> music but also stopped playing following utterances
> which I sent to TTS soon later on after uuid_break.
>
> Anyway I will try it further again and let you all know what I will get.
>
> Cheers,
>
> Xing
>
>



-- 
Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012

We invite research leaders and ambitious early career researchers to 
join us in leading and driving research in key inter-disciplinary themes. 
Please see www.hw.ac.uk/researchleaders for further information and how
to apply.

Heriot-Watt University is a Scottish charity
registered under charity number SC000278.

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From msc at freeswitch.org  Thu Aug  2 20:27:40 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 2 Aug 2012 09:27:40 -0700
Subject: [Freeswitch-users] Freeswitch Auto Retrying a call for 3 times ,
 if it is not answered
In-Reply-To: <1343812553346-7581406.post@n2.nabble.com>
References: <1343812553346-7581406.post@n2.nabble.com>
Message-ID: <CAKzWOxUzckZuEeHUxbVGVhp5WFL0JX6nT+JzWZYnVA1EKRYNBA@mail.gmail.com>

What does it mean that the first user is dialed "three times?" Also, put
this log on pastebin.freeswitch.org and use FreeSWITCH Log as the syntax
highlighting. That makes it much easier to read.

Thanks,
MC

On Wed, Aug 1, 2012 at 2:15 AM, ramesh <ramesh_mind at yahoo.com> wrote:

> Hi Team,
>
> I made a script to dial a user , and if not answered  the second user
> should
> get the dial. Everything works file , Except the first user is dialed for 3
> times , if the user is busy or unanswered , i was trying to figure out what
> could be the problem for this issue, but i couldn't .
>
> Can Anyone Figure out what is the reason for this issue.
> below is my lua dial script
>
>
> session:execute("bridge","{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true}sofia/gateway/
> bandwidth.com/+919xxxxxxxx|sofia/gateway/bandwidth.com/+919xxxxxxxx");
>
>
> following is the log i can get.
>
> Any Help or suggestion would be really helpful!
>
> [NOTICE] sofia.c:5594 Ring-Ready sofia/internal/+919677080275!
> 2012-08-01 07:09:13.212635 [DEBUG] switch_ivr_originate.c:1164 Raw Codec
> Activation Success L16 at 16000hz 1 channel 20ms
> 2012-08-01 07:09:13.212635 [DEBUG] switch_core_codec.c:216
> rtmp/default/+542914850488 Push codec L16:70
> 2012-08-01 07:09:13.212635 [DEBUG] switch_ivr_originate.c:1227 Play
> Ringback
> Tone [%(2000,4000,440,480)]
> 2012-08-01 07:09:16.532655 [DEBUG] switch_core_session.c:875 Send signal
> sofia/internal/+919677080275 [BREAK]
> 2012-08-01 07:09:16.532655 [DEBUG] switch_core_session.c:875 Send signal
> sofia/internal/+919677080275 [BREAK]
> 2012-08-01 07:09:16.552637 [DEBUG] sofia.c:5502 Channel
> sofia/internal/+919677080275 entering state [proceeding][183]
> 2012-08-01 07:09:16.552637 [DEBUG] sofia.c:5513 Remote SDP:
> v=0
> o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14
> s=SIP Media Capabilities
> c=IN IP4 65.115.130.14
> t=0 0
> m=audio 10382 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[SPEEX:99:16000:20:42200]
> 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000]
> 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000]
> 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
> 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:2919 Set Codec
> sofia/internal/+919677080275 PCMU/8000 20 ms 160 samples 64000 bits
> 2012-08-01 07:09:16.552637 [DEBUG] switch_core_codec.c:111
> sofia/internal/+919677080275 Original read codec set to PCMU:0
> 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
> payload to 101
> 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3171 AUDIO RTP
> [sofia/internal/+919677080275] 10.244.15.45 port 30506 -> 65.115.130.14
> port
> 10382 codec: 0 ms: 20
> 2012-08-01 07:09:16.552637 [DEBUG] switch_rtp.c:1661 Starting timer [soft]
> 160 bytes per 20ms
> 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send
> payload to 101
> 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive
> payload to 101
> 2012-08-01 07:09:16.552637 [NOTICE] sofia_glue.c:3945 Pre-Answer
> sofia/internal/+919677080275!
> 2012-08-01 07:09:16.552637 [DEBUG] switch_channel.c:2936
> (sofia/internal/+919677080275) Callstate Change RINGING -> EARLY
> 2012-08-01 07:09:16.552637 [DEBUG] switch_channel.c:2978 Send signal
> rtmp/default/+542914850488 [BREAK]
> 2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2782 Adding tone spec
> 480,620 index 0 hits 1
> 2012-08-01 07:09:16.572648 [DEBUG] switch_core_media_bug.c:457 Attaching
> BUG
> to sofia/internal/+919677080275
> 2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2782 Adding tone spec
> 1776.7 index 1 hits 2
> 2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2877
> sofia/internal/+919677080275 bug already running
> 2012-08-01 07:09:16.572648 [DEBUG] switch_rtp.c:3205 Correct ip/port
> confirmed.
> 2012-08-01 07:09:16.572648 [DEBUG] switch_core_io.c:340 Setting BUG Codec
> PCMU:0
> 2012-08-01 07:09:33.072650 [DEBUG] switch_core_session.c:875 Send signal
> sofia/internal/+919677080275 [BREAK]
> 2012-08-01 07:09:33.072650 [DEBUG] switch_core_session.c:875 Send signal
> sofia/internal/+919677080275 [BREAK]
> 2012-08-01 07:09:33.092641 [DEBUG] sofia.c:5502 Channel
> sofia/internal/+919677080275 entering state [proceeding][183]
> 2012-08-01 07:09:33.092641 [DEBUG] sofia.c:5510 Duplicate SDP
> v=0
> o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14
> s=SIP Media Capabilities
> c=IN IP4 65.115.130.14
> t=0 0
> m=audio 10382 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:2853 Already using PCMU
> 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
> payload to 101
> 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:3141 Audio params are
> unchanged for sofia/internal/+919677080275.
> 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:3151
> sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to
> 0
> 2012-08-01 07:09:36.372634 [DEBUG] switch_core_session.c:875 Send signal
> sofia/internal/+919677080275 [BREAK]
> 2012-08-01 07:09:36.372634 [DEBUG] switch_core_session.c:875 Send signal
> sofia/internal/+919677080275 [BREAK]
> 2012-08-01 07:09:36.392686 [DEBUG] sofia.c:5502 Channel
> sofia/internal/+919677080275 entering state [proceeding][183]
> 2012-08-01 07:09:36.392686 [DEBUG] sofia.c:5510 Duplicate SDP
> v=0
> o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14
> s=SIP Media Capabilities
> c=IN IP4 65.115.130.14
> t=0 0
> m=audio 10382 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:2853 Already using PCMU
> 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
> payload to 101
> 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:3141 Audio params are
> unchanged for sofia/internal/+919677080275.
> 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:3151
> sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to
> 0
> 2012-08-01 07:09:38.512638 [DEBUG] libdingaling.c:1624 Sent keep alive
> signal
> 2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal
> sofia/internal/+919677080275 [BREAK]
> 2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal
> sofia/internal/+919677080275 [BREAK]
> 2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal
> sofia/internal/+919677080275 [BREAK]
> 2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal
> sofia/internal/+919677080275 [BREAK]
> 2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5502 Channel
> sofia/internal/+919677080275 entering state [proceeding][183]
> 2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5510 Duplicate SDP
> v=0
> o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14
> s=SIP Media Capabilities
> c=IN IP4 65.115.130.14
> t=0 0
> m=audio 10382 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:2853 Already using PCMU
> 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
> payload to 101
> 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3141 Audio params are
> unchanged for sofia/internal/+919677080275.
> 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3151
> sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to
> 0
> 2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5502 Channel
> sofia/internal/+919677080275 entering state [proceeding][183]
> 2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5510 Duplicate SDP
> v=0
> o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14
> s=SIP Media Capabilities
> c=IN IP4 65.115.130.14
> t=0 0
> m=audio 10382 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4798 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:2853 Already using PCMU
> 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send
> payload to 101
> 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3141 Audio params are
> unchanged for sofia/internal/+919677080275.
> 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3151
> sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to
> 0
> 2012-08-01 07:10:11.012635 [DEBUG] switch_core_codec.c:241
> rtmp/default/+542914850488 Restore previous codec SPEEX:99.
> 2012-08-01 07:10:11.012635 [DEBUG] switch_channel.c:2852
> (sofia/internal/+919677080275) Callstate Change EARLY -> HANGUP
> 2012-08-01 07:10:11.012635 [NOTICE] switch_ivr_originate.c:3182 Hangup
> sofia/internal/+919677080275 [CS_CONSUME_MEDIA] [NO_ANSWER]
>
>
> Thanks
> Ramesh
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Auto-Retrying-a-call-for-3-times-if-it-is-not-answered-tp7581406.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From philq at qsystemsengineering.com  Thu Aug  2 20:35:04 2012
From: philq at qsystemsengineering.com (Phil Quesinberry)
Date: Thu, 02 Aug 2012 12:35:04 -0400
Subject: [Freeswitch-users] Conditional testing on caller_id_name
Message-ID: <016501cd70cc$c7e4c3b0$57ae4b10$@com>

Thanks Brian,
 
I think the lights just came on...
 
I think I finally see what's going on here.  Looking through the wiki some
more, I came across the following:
"The XML Dialplan has the ability to test a number of conditions based upon
variables with expressions; however, it needs to be understood that some
variables may not be available for conditional testing until the first
transfer or execute_extension is performed."
 
When I do a conditional test such as the following:
<condition field="caller_id_name" expression="^Currently running a lookup">

the console output shows the ani instead of the CNAM in parentheses as
shown, and fails:
Dialplan: sofia/external/4105551212 at 140.239.xx.xx Regex (FAIL)
[currently_running] caller_id_name(4105551212) =~ /^Currently running a
lookup/ break=on-false

BUT when I ask for the variable as follows:
<action application="log" data="${caller_id_name}"/>

I get the expected value in the console output:
EXECUTE sofia/external/4105551212 at 140.239.xx.xx log(SMITH,JOHN)
2012-08-02 12:12:50.679592 [DEBUG] mod_dptools.c:1458 SMITH,JOHN

(the names and numbers have been changed to protect the clueless)

So just to be sure, is this in fact what's going on here?  I learned
something today... now that I understand it, I can deal with it.  By next
year I'm bound to comprehend things well enough to get something useful from
ClueCon.

Thanks!


The rest of the message that follows was written before I came across this
little tidbit of information but I'm keeping it in here for reference:
 
<clueless banter follows>
I have no idea how to use Perl in FS but after a bit of looking around it
appears that I would need to do something like:
<action application="perl" data="/usr/src/freeswitch/scripts/perl/cnam.cgi
$1"/>
 
Is that correct?
 
Better yet, is there any way to get to the CID name variable without having
to resort to Perl?  Doing an 'info' in the dialplan during an incoming call
shows the following:
Caller-Caller-ID-Name: [JOHN SMITH]
 
How can I get to that data?  'caller_id_name' just seems to contain the
number instead of the CNAM info, which doesn't make any sense.
 
The wiki shows the following example to set a variable directly:
<action application="set"
data="cid_name=${cidlookup(${caller_id_number})}"/>    and this looks like
it might work, but I'd like to understand why the variables aren't already
set after the cidlookup is done in public.xml.
<clueless banter ends/>
 
Phil Quesinberry
Q Systems Engineering, Inc.
Electronic Controls and Embedded Systems Development
(410) 969-8002
http://www.qsystemsengineering.com

----------
 
[Freeswitch-users] Setting effecting_caller_id_name
Brian West 
Sat Jul 28 17:46:14 MSD 2012
 
Sounds like your doing openCNAM, check the cnam.cgi in tree, you should be
checking the response code and not the text returned.  I built the wrapper
to clean up the input before I query because of this.
 
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: +1.213.286.0410  |  F: +1.213.286.0401  |  M: +1.918.424.WEST
iNUM: +883 5100 1286 0410
 
On Jul 27, 2012, at 10:06 AM, Ken Rice <krice at freeswitch.org> wrote:
 
Re: [Freeswitch-users] Setting effecting_caller_id_name This exmaple should
be correct provided that caller_id_name containts "Currently runnint a
lookup" with this pinned to that start of the field...
 
On that extension in your dialplan toss a <anti-action application="info"/>
 
If your regex isnt matching on the caller_id_name field the above will show
you whats in there (along with every thing else too heh)
 
K
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From msc at freeswitch.org  Thu Aug  2 20:42:21 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 2 Aug 2012 09:42:21 -0700
Subject: [Freeswitch-users] Conditional testing on caller_id_name
In-Reply-To: <016501cd70cc$c7e4c3b0$57ae4b10$@com>
References: <016501cd70cc$c7e4c3b0$57ae4b10$@com>
Message-ID: <CAKzWOxUZfPd_n18qSAZCAqDnSUtmwTz-euvgK+UJppMjsQU=ow@mail.gmail.com>

Phil,

Perhaps you could try running the action inline? Change:
<action application="set"
data="cid_name=${cidlookup(${caller_id_number})}"/>

to:
<action application="set" data="cid_name=${cidlookup(${caller_id_number})}"
inline="true"/>

And see what happens...

-Michael

On Thu, Aug 2, 2012 at 9:35 AM, Phil Quesinberry <
philq at qsystemsengineering.com> wrote:

> **
>
> Thanks Brian,
>
>
>
> I think the lights just came on...
>
>
>
> I think I finally see what?s going on here.  Looking through the wikisome more,
> I came across the following:
>
> ?The XML Dialplan has the ability to test a number of conditions based
> upon variables with expressions; however, it needs to be understood that
> some variables may not be available for conditional testing until the first
> transfer or execute_extension is performed.?
>
>
>
> When I do a conditional test such as the following:
>
> <condition field="caller_id_name" expression="^Currently running a lookup">
>
> the console output shows the ani instead of the CNAM in parentheses as shown,
> and fails:
>
> Dialplan: sofia/external/4105551212 at 140.239.xx.xx Regex (FAIL)
> [currently_running] caller_id_name(4105551212) =~ /^Currently running a
> lookup/ break=on-false
>
> BUT when I ask for the variable as follows:
>
> <action application="log" data="${caller_id_name}"/>
>
> I get the expected value in the console output:
>
> EXECUTE sofia/external/4105551212 at 140.239.xx.xx log(SMITH,JOHN)
>
> 2012-08-02 12:12:50.679592 [DEBUG] mod_dptools.c:1458 SMITH,JOHN
>
> (the names and numbers have been changed to protect the clueless)
>
> So just to be sure, is this in fact what's going on here?  I learned
> something today... now that I understand it, I can deal with it.  By next
> year I'm bound to comprehend things well enough to get something useful
> from ClueCon.
>
> Thanks!
>
> The rest of the message that follows was written before I came across
> this little tidbit of information but I?m keeping it in here for reference:
>
>
>
> <clueless banter follows>
>
> I have no idea how to use Perl in FS but after a bit of looking around it
> appears that I would need to do something like:
>
> <action application="perl" data="/usr/src/freeswitch/scripts/perl/cnam.cgi
> $1"/>
>
>
>
> Is that correct?
>
>
>
> Better yet, is there any way to get to the CID name variable without
> having to resort to Perl?  Doing an ?info? in the dialplan during an
> incoming call shows the following:
>
> Caller-Caller-ID-Name: [JOHN SMITH]
>
>
>
> How can I get to that data?  ?caller_id_name? just seems to contain the
> number instead of the CNAM info, which doesn?t make any sense.
>
>
>
> The wiki shows the following example to set a variable directly:
>
> <action application="set"
> data="cid_name=${cidlookup(${caller_id_number})}"/>    and this looks like
> it might work, but I?d like to understand why the variables aren?t already
> set after the cidlookup is done in public.xml.
>
> <clueless banter ends/>
>
>
>
> Phil Quesinberry
>
> Q Systems Engineering, Inc.
>
> Electronic Controls and Embedded Systems Development
>
> (410) 969-8002
>
> *http://www.qsystemsengineering.co**m*<http://www.qsystemsengineering.com>
>
> ----------
>
>
>
> [Freeswitch-users] Setting effecting_caller_id_name
>
> Brian West
>
> Sat Jul 28 17:46:14 MSD 2012
>
>
>
> Sounds like your doing openCNAM, check the cnam.cgi in tree, you should be
>
> checking the response code and not the text returned.  I built the wrapper
>
> to clean up the input before I query because of this.
>
>
>
> --
>
> Brian West
>
> brian at freeswitch.org
>
> FreeSWITCH Solutions, LLC
>
> PO BOX PO BOX 2531
>
> Brookfield, WI 53008-2531
>
> Twitter: @FreeSWITCH_Wire
>
> T: +1.213.286.0410  |  F: +1.213.286.0401  |  M: +1.918.424.WEST
>
> iNUM: +883 5100 1286 0410
>
>
>
> On Jul 27, 2012, at 10:06 AM, Ken Rice <krice at freeswitch.org> wrote:
>
>
>
> Re: [Freeswitch-users] Setting effecting_caller_id_name This exmaple should
>
> be correct provided that caller_id_name containts ?Currently runnint a
>
> lookup? with this pinned to that start of the field...
>
>
>
> On that extension in your dialplan toss a <anti-action application=?info?/>
>
>
>
> If your regex isnt matching on the caller_id_name field the above will show
>
> you whats in there (along with every thing else too heh)
>
>
>
> K
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From avi at avimarcus.net  Thu Aug  2 20:45:36 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 2 Aug 2012 19:45:36 +0300
Subject: [Freeswitch-users] Conditional testing on caller_id_name
In-Reply-To: <016501cd70cc$c7e4c3b0$57ae4b10$@com>
References: <016501cd70cc$c7e4c3b0$57ae4b10$@com>
Message-ID: <CALNZuwn1xHM0Wev_bsgaS9pQyBBMQwt4HP1RKTYmVjnUUbuRdQ@mail.gmail.com>

Short snippet of info:
e.g. caller_id_number is from the
caller_profile<http://wiki.freeswitch.org/wiki/Caller_profile_field>
which
doesn't really get modified.
${vars} are vars that do get modified.

-Avi
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From lists at telefaks.de  Thu Aug  2 20:54:11 2012
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 02 Aug 2012 18:54:11 +0200
Subject: [Freeswitch-users] mod_fifo: Agent in multiple fifos gets multiple
	calls at once
Message-ID: <501AB0B3.4090009@telefaks.de>

Hello,

we have multiple fifos with multiple agents. Some agents serve multiple
fifos.
We currently see the following behaviour, which causes unintended hangups:

  * Agent A1 gets a call from caller C1 in fifo F1
  * Agent A1 is now in a call with C1
  * Another call is coming in from caller C2 to fifo F2 which is also
    served by A1
  * Freeswitch is trying the place the call C2 to Agent A1 although A1
    is in a call

Normally this is not a problem for us, but sometimes Freeswitch tries to
place almost simultaneously 2 calls from 2 different fifos to the same
agent (I have a wireshark example of 2 invites within 1 msec). This
causes our Aastra phones to hangup the 1st call calls when Freeswitch
sends a CANCEL for the second call to the phone.

My question: Are Fifos working independently in Freeswitch or is there a
way to share the agent's phone status across several fifos?

-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de


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From mario_fs at mgtech.com  Thu Aug  2 20:50:36 2012
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 2 Aug 2012 09:50:36 -0700
Subject: [Freeswitch-users] How to file FreeSwitch Enhancements requests?
Message-ID: <D6CCBDDC-EC0A-46FC-92F1-35177A3BDB5B@mgtech.com>

Is there an official way to suggest FreeSwitch enhancements? Where should it go and how much doc/explanation is needed? In this case:

1. in bridge: Add the ability to specify variables that apply only to one group_call but all users in that group.

2. In bridge: Add the ability to allow leg_delay_start, etc. to work using enterprise syntax.

I posted this yesterday which may explain more: (has gotten no response)


What I need: to have different variable/settings for each target group in a bridge command, and allow leg_delay_start to work. It's related to trying to fix other issues and holding up progress for them. Thanks for any help,
Mario G

I scoured the wiki and tried many things. Normally, enterprise syntax would solve this, but there is a catch: you can't use leg_delay_start which I need, it is ignored when using enterprise. I found that using brackets [] only applies the variable to the first user in the group, not the whole group so that won't work.

What I have now but need to have alert_info moved/apply to the mgt group only: 
<action application="bridge" data="{originate_timeout=45,alert_info=n=${lua_ringtone}}${group_call(mgtbria@${domain_name}+A)},${group_call(mgt@${domain_name}+A)},[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551212,[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551313"/>
    

 This solves variables for groups but breaks leg_delay_start, etc.:
<action application="bridge" data="<originate_timeout=45>}${group_call(mgtbria@${domain_name}+A)}:_:{alert_info=n=${lua_ringtone}${group_call(mgt@${domain_name}+A)}:_:[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551212,[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551313"/> 
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From leonardo at daitangroup.com  Thu Aug  2 20:58:20 2012
From: leonardo at daitangroup.com (Leonardo)
Date: Thu, 2 Aug 2012 13:58:20 -0300
Subject: [Freeswitch-users] How to match a message sent using chat api with
	MESSAGE event
Message-ID: <501AB1AC.8040303@daitangroup.com>

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From msc at freeswitch.org  Thu Aug  2 21:05:05 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 2 Aug 2012 10:05:05 -0700
Subject: [Freeswitch-users] How to file FreeSwitch Enhancements requests?
In-Reply-To: <D6CCBDDC-EC0A-46FC-92F1-35177A3BDB5B@mgtech.com>
References: <D6CCBDDC-EC0A-46FC-92F1-35177A3BDB5B@mgtech.com>
Message-ID: <CAKzWOxVrVn_cQ1brYhvF7g4ZYdqq3PtsRLJ+37YWN65wKPQ1QA@mail.gmail.com>

You can do this with a Jira. Also, after you open the Jira go ahead and put
up a bounty on the wiki:
http://wiki.freeswitch.org/wiki/Bounty

Sometimes that will grease the wheels.

-Michael

On Thu, Aug 2, 2012 at 9:50 AM, Mario G <mario_fs at mgtech.com> wrote:

> Is there an official way to suggest FreeSwitch enhancements? Where should
> it go and how much doc/explanation is needed? In this case:
>
> 1. in bridge: Add the ability to specify variables that apply only to one
> group_call but all users in that group.
>
> 2. In bridge: Add the ability to allow leg_delay_start, etc. to work using
> enterprise syntax.
>
> I posted this yesterday which may explain more: (has gotten no response)
>
>
> What I need: to have different variable/settings for each target group in
> a bridge command, and allow leg_delay_start to work. It's related to trying
> to fix other issues and holding up progress for them. Thanks for any help,
> Mario G
>
> I scoured the wiki and tried many things. Normally, enterprise syntax
> would solve this, but there is a catch: you can't use leg_delay_start which
> I need, it is ignored when using enterprise. I found that using brackets []
> only applies the variable to the first user in the group, not the whole
> group so that won't work.
>
> What I have now but need to have alert_info moved/apply to the mgt group
> only:
> <action application="bridge" data=
> "{originate_timeout=45,alert_info=n=${lua_ringtone}}${group_call(mgtbria@
> ${domain_name}+A)},${group_call(mgt@
> ${domain_name}+A)},[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/
> 19165551212
> ,[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/
> 19165551313"/>
>
>
>
>  This solves variables for groups but breaks leg_delay_start, etc.:
> <action application="bridge" data=
> "<originate_timeout=45>}${group_call(mgtbria@${domain_name}+A)}:_:
> {alert_info=n=${lua_ringtone}${group_call(mgt@
> ${domain_name}+A)}:_:[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/
> 19165551212
> ,[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/
> 19165551313"/>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From m at edisonjunior.com  Thu Aug  2 21:15:29 2012
From: m at edisonjunior.com (Mark Dillon)
Date: Thu, 2 Aug 2012 10:15:29 -0700
Subject: [Freeswitch-users] queue_dtmf and send_dtmf merging digits
Message-ID: <CAFucMnUs8mgSvH-q_SZmYdqgfA4uNWy5hzF-y7HAVAW+UrA1hA@mail.gmail.com>

Hello. I am sending outbound dtmf over g729 using rfc2833. Generally it
works perfectly. The only issue is that, when 2 subsequent digits are the
same, the receiver detects them as a single digit. For instance, if I send
1234, it is detected as expected as 1234. However, if I change that to
2234, it is detected as 234.

I have played with rtp-digit-delay and dtmf-duration extensively to no
avail. I've also tried both queue_dtmf and execute_on_answer=send_dtmf.
Anyone have any ideas? Any help much appreciated. Here is the extension
that I'm using in my dialplan:

  <extension name="test_conference">
    <condition field="destination_number" expression="^test_conference$">
      <action application="playback" data="wait.wav" />
      <action application="set"
data="effective_caller_id_number=5555555555"/>
      <action application="export"
data="nolocal:absolute_codec_string=G729"/>
      <action application="flush_dtmf"/>
      <action application="queue_dtmf" data="2234"/>
      <!-- <action application="export"
data="nolocal:execute_on_answer=send_dtmf 2234" />-->
      <action application="bridge"
data="sofia/internal/${bridge_number}@${trunk_ip}"
/>
      <action application="hangup" />
    </condition>
  </extension>

Thanks,
Mark
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From mario_fs at mgtech.com  Thu Aug  2 21:05:31 2012
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 2 Aug 2012 10:05:31 -0700
Subject: [Freeswitch-users] Anyine have Bria iOS tipes for FreeSwitch?
Message-ID: <23359C2E-5614-4500-8BCC-DC84A0383497@mgtech.com>

I am at my wits end after several months of Bria problems. Counterpath keeps pointing the finger at FreeSwitch. I am wondering if anyone here has Bria working well and if they can share tips user definitions? Bria is set for TCP to reduce battery usage. Thanks.
Mario G



From bdfoster at endigotech.com  Thu Aug  2 21:24:41 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 2 Aug 2012 13:24:41 -0400
Subject: [Freeswitch-users] mod_fifo: Agent in multiple fifos gets
 multiple calls at once
In-Reply-To: <501AB0B3.4090009@telefaks.de>
References: <501AB0B3.4090009@telefaks.de>
Message-ID: <CAKV-jFxQVRVekjHBQAXFQYZuobXb3DVAUPOdxc49+m=7QO5rOw@mail.gmail.com>

isn't this covered in mod_callcenter?

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 2, 2012 12:57 PM, "Peter Steinbach" <lists at telefaks.de> wrote:

>  Hello,
>
> we have multiple fifos with multiple agents. Some agents serve multiple
> fifos.
> We currently see the following behaviour, which causes unintended hangups:
>
>    - Agent A1 gets a call from caller C1 in fifo F1
>    - Agent A1 is now in a call with C1
>    - Another call is coming in from caller C2 to fifo F2 which is also
>    served by A1
>     - Freeswitch is trying the place the call C2 to Agent A1 although A1
>    is in a call
>
> Normally this is not a problem for us, but sometimes Freeswitch tries to
> place almost simultaneously 2 calls from 2 different fifos to the same
> agent (I have a wireshark example of 2 invites within 1 msec). This causes
> our Aastra phones to hangup the 1st call calls when Freeswitch sends a
> CANCEL for the second call to the phone.
>
> My question: Are Fifos working independently in Freeswitch or is there a
> way to share the agent's phone status across several fifos?
>
>  --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Aug  2 22:00:20 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 2 Aug 2012 11:00:20 -0700
Subject: [Freeswitch-users] The FreeSWITCH Devs - Let's Buy 'Em a Drink!
Message-ID: <CAKzWOxXrfUsacnNc0SexogND=y8-oqRhZb5rYwaRXbKEsRRn8A@mail.gmail.com>

Hey all!

It's ClueCon time and we'll all be in Chicago next week. As a token of
appreciation for all of their hard work we'd like to invite everyone to
donate a few dollars (or more!) in order to buy Anthony Minessale, Brian
West, and Mike Jerris a very well-deserved drink while they're in Chicago.
Simply click the Donate link on www.FreeSWITCH.org and add a note to seller
that this is for buying the developers a drink.

On a personal note I would just like to say that I have seen first hand
just how hard-working and dedicated these gentlemen really are. <hat tip/>
Thank you for making the FreeSWITCH project so awesome!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From david.villasmil.work at gmail.com  Thu Aug  2 22:01:41 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 2 Aug 2012 20:01:41 +0200
Subject: [Freeswitch-users] ESL Log to console
In-Reply-To: <CAKzWOxWT5J=HFJ8hB=bpo32ACtmmqrNYqNGSar=2incVFWej+w@mail.gmail.com>
References: <CAFGRPVpu_3t1JG8SPqEb8J9JBn6VYOXL1Aa79u_7Hj-uO0WB0A@mail.gmail.com>
	<CAKzWOxWT5J=HFJ8hB=bpo32ACtmmqrNYqNGSar=2incVFWej+w@mail.gmail.com>
Message-ID: <CAFGRPVrV1UUunYtVs68+EX_PXn-yDGxEZhN3AJEBJY_KYoDcCg@mail.gmail.com>

Hello Michael,

Yes, there's a call in progress. Executing it like you said worked. (api)


Now, I'm trying to set a variable like so:

$con->execute("set","customer_company             = " .  $local_variable ,
$uuid );

But it doesn't get set...

:(

Thanks for your help!

David


On Thu, Aug 2, 2012 at 6:25 PM, Michael Collins <msc at freeswitch.org> wrote:

> David,
>
> Do you have a call in progress at this point? If not then you'll need to
> supply a uuid of a live call, as mentioned here:
> http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute
>
> Remember, "execute" means "execute dialplan application" so if there's no
> channel then "execute" really doesn't mean a whole lot.
>
> Alternatively you could try something like this:
>
> $con->api("log","WARNING Don't cross the streams!!");
>
> Remember this rule of thumb: pretty much anything you type at fs_cli is an
> "API" and therefore you can use $con->api(), whereas anything that is a
> diaplan application requires an actual channel on which to run
> $con->execute().
>
> Hope that makes sense... :)
>
> -Michael
>
> On Thu, Aug 2, 2012 at 8:43 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello Guys,
>>
>> I'm starting off with ESL, which is cool, but I'm trying to log to the
>> console like Lua's "freeswitch.consoleLog("info", "BLAH BLAH");" doing it
>> like:
>>
>> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>> $con->execute("log", "1, BlahBlah");
>>
>>
>> But nothing gets in the log files or console... and I can't find any
>> documentation as to how to log using "execute"...
>>
>> any ideas?
>>
>> Thanks!
>>
>> David
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Thu Aug  2 22:04:43 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 02 Aug 2012 13:04:43 -0500
Subject: [Freeswitch-users] Anyine have Bria iOS tipes for FreeSwitch?
In-Reply-To: <23359C2E-5614-4500-8BCC-DC84A0383497@mgtech.com>
Message-ID: <CC402B6B.3ACBF%krice@freeswitch.org>

What kind of problems are you having? I use Bria on my iPad all the time and
don't seem to have issues with it


On 8/2/12 12:05 PM, "Mario G" <mario_fs at mgtech.com> wrote:

> I am at my wits end after several months of Bria problems. Counterpath keeps
> pointing the finger at FreeSwitch. I am wondering if anyone here has Bria
> working well and if they can share tips user definitions? Bria is set for TCP
> to reduce battery usage. Thanks.
> Mario G
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Aug  2 22:19:34 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 2 Aug 2012 11:19:34 -0700
Subject: [Freeswitch-users] ESL Log to console
In-Reply-To: <CAFGRPVrV1UUunYtVs68+EX_PXn-yDGxEZhN3AJEBJY_KYoDcCg@mail.gmail.com>
References: <CAFGRPVpu_3t1JG8SPqEb8J9JBn6VYOXL1Aa79u_7Hj-uO0WB0A@mail.gmail.com>
	<CAKzWOxWT5J=HFJ8hB=bpo32ACtmmqrNYqNGSar=2incVFWej+w@mail.gmail.com>
	<CAFGRPVrV1UUunYtVs68+EX_PXn-yDGxEZhN3AJEBJY_KYoDcCg@mail.gmail.com>
Message-ID: <CAKzWOxU+nY6Q-xfWd2bd8EUQYsefMpgSeR6-gDH=qx8+Vg4QiA@mail.gmail.com>

Can't use execute and set unless this is an outbound ESL connection, which
it seems not to be. I'd try this:

$con->api("uuid_setvar","$uuid customer_company $local_variable");

-MC

On Thu, Aug 2, 2012 at 11:01 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello Michael,
>
> Yes, there's a call in progress. Executing it like you said worked. (api)
>
>
> Now, I'm trying to set a variable like so:
>
> $con->execute("set","customer_company             = " .  $local_variable ,
> $uuid );
>
> But it doesn't get set...
>
> :(
>
> Thanks for your help!
>
> David
>
>
> On Thu, Aug 2, 2012 at 6:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> David,
>>
>> Do you have a call in progress at this point? If not then you'll need to
>> supply a uuid of a live call, as mentioned here:
>> http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute
>>
>> Remember, "execute" means "execute dialplan application" so if there's no
>> channel then "execute" really doesn't mean a whole lot.
>>
>> Alternatively you could try something like this:
>>
>> $con->api("log","WARNING Don't cross the streams!!");
>>
>> Remember this rule of thumb: pretty much anything you type at fs_cli is
>> an "API" and therefore you can use $con->api(), whereas anything that is a
>> diaplan application requires an actual channel on which to run
>> $con->execute().
>>
>> Hope that makes sense... :)
>>
>> -Michael
>>
>> On Thu, Aug 2, 2012 at 8:43 AM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello Guys,
>>>
>>> I'm starting off with ESL, which is cool, but I'm trying to log to the
>>> console like Lua's "freeswitch.consoleLog("info", "BLAH BLAH");" doing it
>>> like:
>>>
>>> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>>> $con->execute("log", "1, BlahBlah");
>>>
>>>
>>> But nothing gets in the log files or console... and I can't find any
>>> documentation as to how to log using "execute"...
>>>
>>> any ideas?
>>>
>>> Thanks!
>>>
>>> David
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From david.villasmil.work at gmail.com  Thu Aug  2 22:25:38 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 2 Aug 2012 20:25:38 +0200
Subject: [Freeswitch-users] ESL Log to console
In-Reply-To: <CAKzWOxU+nY6Q-xfWd2bd8EUQYsefMpgSeR6-gDH=qx8+Vg4QiA@mail.gmail.com>
References: <CAFGRPVpu_3t1JG8SPqEb8J9JBn6VYOXL1Aa79u_7Hj-uO0WB0A@mail.gmail.com>
	<CAKzWOxWT5J=HFJ8hB=bpo32ACtmmqrNYqNGSar=2incVFWej+w@mail.gmail.com>
	<CAFGRPVrV1UUunYtVs68+EX_PXn-yDGxEZhN3AJEBJY_KYoDcCg@mail.gmail.com>
	<CAKzWOxU+nY6Q-xfWd2bd8EUQYsefMpgSeR6-gDH=qx8+Vg4QiA@mail.gmail.com>
Message-ID: <CAFGRPVqOfe-507gV1JwBGWc2BijFp50-Vr8_u2QzM8d8DrfS=Q@mail.gmail.com>

Ouch!

Ok, this is not like in Lua... Ok, I'll try that.... is there any
documentation regarding what can and can't be set and the conditions for
setting them? What about bridging channels, etc?

:(

David

On Thu, Aug 2, 2012 at 8:19 PM, Michael Collins <msc at freeswitch.org> wrote:

> Can't use execute and set unless this is an outbound ESL connection, which
> it seems not to be. I'd try this:
>
> $con->api("uuid_setvar","$uuid customer_company $local_variable");
>
> -MC
>
>
> On Thu, Aug 2, 2012 at 11:01 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello Michael,
>>
>> Yes, there's a call in progress. Executing it like you said worked. (api)
>>
>>
>> Now, I'm trying to set a variable like so:
>>
>> $con->execute("set","customer_company             = " .  $local_variable
>> , $uuid );
>>
>> But it doesn't get set...
>>
>> :(
>>
>> Thanks for your help!
>>
>> David
>>
>>
>> On Thu, Aug 2, 2012 at 6:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> David,
>>>
>>> Do you have a call in progress at this point? If not then you'll need to
>>> supply a uuid of a live call, as mentioned here:
>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute
>>>
>>> Remember, "execute" means "execute dialplan application" so if there's
>>> no channel then "execute" really doesn't mean a whole lot.
>>>
>>> Alternatively you could try something like this:
>>>
>>> $con->api("log","WARNING Don't cross the streams!!");
>>>
>>> Remember this rule of thumb: pretty much anything you type at fs_cli is
>>> an "API" and therefore you can use $con->api(), whereas anything that is a
>>> diaplan application requires an actual channel on which to run
>>> $con->execute().
>>>
>>> Hope that makes sense... :)
>>>
>>> -Michael
>>>
>>> On Thu, Aug 2, 2012 at 8:43 AM, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> Hello Guys,
>>>>
>>>> I'm starting off with ESL, which is cool, but I'm trying to log to the
>>>> console like Lua's "freeswitch.consoleLog("info", "BLAH BLAH");" doing it
>>>> like:
>>>>
>>>> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>>>> $con->execute("log", "1, BlahBlah");
>>>>
>>>>
>>>> But nothing gets in the log files or console... and I can't find any
>>>> documentation as to how to log using "execute"...
>>>>
>>>> any ideas?
>>>>
>>>> Thanks!
>>>>
>>>> David
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Michael S Collins
>>> Twitter: @mercutioviz
>>> http://www.FreeSWITCH.org
>>> http://www.ClueCon.com
>>> http://www.OSTAG.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sazzadbinkamal at gmail.com  Thu Aug  2 22:28:25 2012
From: sazzadbinkamal at gmail.com (Sazzad)
Date: Fri, 3 Aug 2012 00:28:25 +0600
Subject: [Freeswitch-users] FreeSwitch Error: "username param is REQUIRED!"
Message-ID: <CADyf1QM87w4hQByq8nz2ujeH_z9a5+R43aJmG4Prmpg=M3d4ag@mail.gmail.com>

Hi!

I'm trying out FreeSwitch with sample configuration at
sip_profile/external/example.xml. I removed few required line from there
and issued command:

sofia profile external restart reloadxml


Its giving me the following error:

[ERR] sofia.c:2606 ERROR: username param is REQUIRED!



>From the source code I see at the specified line checks if "username" is
null, if so we get this error. How come I get this error while I see a
string is there in the configuration file?

-- 
Sincerely,
Sazzad Bin Kamal
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From msc at freeswitch.org  Thu Aug  2 23:07:18 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 2 Aug 2012 12:07:18 -0700
Subject: [Freeswitch-users] ESL Log to console
In-Reply-To: <CAFGRPVqOfe-507gV1JwBGWc2BijFp50-Vr8_u2QzM8d8DrfS=Q@mail.gmail.com>
References: <CAFGRPVpu_3t1JG8SPqEb8J9JBn6VYOXL1Aa79u_7Hj-uO0WB0A@mail.gmail.com>
	<CAKzWOxWT5J=HFJ8hB=bpo32ACtmmqrNYqNGSar=2incVFWej+w@mail.gmail.com>
	<CAFGRPVrV1UUunYtVs68+EX_PXn-yDGxEZhN3AJEBJY_KYoDcCg@mail.gmail.com>
	<CAKzWOxU+nY6Q-xfWd2bd8EUQYsefMpgSeR6-gDH=qx8+Vg4QiA@mail.gmail.com>
	<CAFGRPVqOfe-507gV1JwBGWc2BijFp50-Vr8_u2QzM8d8DrfS=Q@mail.gmail.com>
Message-ID: <CAKzWOxWS98-B5q1rVO3eE0GR-i=PtO-biHDgDCULmOE9n5eRYw@mail.gmail.com>

Well, it all depends on how you're calling this script. Is it being called
from the dialplan?
-MC

On Thu, Aug 2, 2012 at 11:25 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Ouch!
>
> Ok, this is not like in Lua... Ok, I'll try that.... is there any
> documentation regarding what can and can't be set and the conditions for
> setting them? What about bridging channels, etc?
>
> :(
>
> David
>
> On Thu, Aug 2, 2012 at 8:19 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Can't use execute and set unless this is an outbound ESL connection,
>> which it seems not to be. I'd try this:
>>
>> $con->api("uuid_setvar","$uuid customer_company $local_variable");
>>
>> -MC
>>
>>
>> On Thu, Aug 2, 2012 at 11:01 AM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello Michael,
>>>
>>> Yes, there's a call in progress. Executing it like you said worked. (api)
>>>
>>>
>>> Now, I'm trying to set a variable like so:
>>>
>>> $con->execute("set","customer_company             = " .  $local_variable
>>> , $uuid );
>>>
>>> But it doesn't get set...
>>>
>>> :(
>>>
>>> Thanks for your help!
>>>
>>> David
>>>
>>>
>>> On Thu, Aug 2, 2012 at 6:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> David,
>>>>
>>>> Do you have a call in progress at this point? If not then you'll need
>>>> to supply a uuid of a live call, as mentioned here:
>>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute
>>>>
>>>> Remember, "execute" means "execute dialplan application" so if there's
>>>> no channel then "execute" really doesn't mean a whole lot.
>>>>
>>>> Alternatively you could try something like this:
>>>>
>>>> $con->api("log","WARNING Don't cross the streams!!");
>>>>
>>>> Remember this rule of thumb: pretty much anything you type at fs_cli is
>>>> an "API" and therefore you can use $con->api(), whereas anything that is a
>>>> diaplan application requires an actual channel on which to run
>>>> $con->execute().
>>>>
>>>> Hope that makes sense... :)
>>>>
>>>> -Michael
>>>>
>>>> On Thu, Aug 2, 2012 at 8:43 AM, David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>>> Hello Guys,
>>>>>
>>>>> I'm starting off with ESL, which is cool, but I'm trying to log to the
>>>>> console like Lua's "freeswitch.consoleLog("info", "BLAH BLAH");" doing it
>>>>> like:
>>>>>
>>>>> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>>>>> $con->execute("log", "1, BlahBlah");
>>>>>
>>>>>
>>>>> But nothing gets in the log files or console... and I can't find any
>>>>> documentation as to how to log using "execute"...
>>>>>
>>>>> any ideas?
>>>>>
>>>>> Thanks!
>>>>>
>>>>> David
>>>>>
>>>>>
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From msc at freeswitch.org  Thu Aug  2 23:11:30 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 2 Aug 2012 12:11:30 -0700
Subject: [Freeswitch-users] FreeSwitch Error: "username param is
	REQUIRED!"
In-Reply-To: <CADyf1QM87w4hQByq8nz2ujeH_z9a5+R43aJmG4Prmpg=M3d4ag@mail.gmail.com>
References: <CADyf1QM87w4hQByq8nz2ujeH_z9a5+R43aJmG4Prmpg=M3d4ag@mail.gmail.com>
Message-ID: <CAKzWOxUzquwprRcH_M-xsE8yFpP=sx7AnqY7mZ+h3KWZVmNOLw@mail.gmail.com>

Long story short: just put in a dummy value if you are not using SIP
authentication for this gateway.
-MC

On Thu, Aug 2, 2012 at 11:28 AM, Sazzad <sazzadbinkamal at gmail.com> wrote:

> Hi!
>
> I'm trying out FreeSwitch with sample configuration at
> sip_profile/external/example.xml. I removed few required line from there
> and issued command:
>
> sofia profile external restart reloadxml
>
>
> Its giving me the following error:
>
> [ERR] sofia.c:2606 ERROR: username param is REQUIRED!
>
>
>
> From the source code I see at the specified line checks if "username" is
> null, if so we get this error. How come I get this error while I see a
> string is there in the configuration file?
>
> --
> Sincerely,
> Sazzad Bin Kamal
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From david.villasmil.work at gmail.com  Thu Aug  2 23:21:29 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 2 Aug 2012 21:21:29 +0200
Subject: [Freeswitch-users] ESL Log to console
In-Reply-To: <CAKzWOxWS98-B5q1rVO3eE0GR-i=PtO-biHDgDCULmOE9n5eRYw@mail.gmail.com>
References: <CAFGRPVpu_3t1JG8SPqEb8J9JBn6VYOXL1Aa79u_7Hj-uO0WB0A@mail.gmail.com>
	<CAKzWOxWT5J=HFJ8hB=bpo32ACtmmqrNYqNGSar=2incVFWej+w@mail.gmail.com>
	<CAFGRPVrV1UUunYtVs68+EX_PXn-yDGxEZhN3AJEBJY_KYoDcCg@mail.gmail.com>
	<CAKzWOxU+nY6Q-xfWd2bd8EUQYsefMpgSeR6-gDH=qx8+Vg4QiA@mail.gmail.com>
	<CAFGRPVqOfe-507gV1JwBGWc2BijFp50-Vr8_u2QzM8d8DrfS=Q@mail.gmail.com>
	<CAKzWOxWS98-B5q1rVO3eE0GR-i=PtO-biHDgDCULmOE9n5eRYw@mail.gmail.com>
Message-ID: <CAFGRPVreuL_rZRo4D=uNKXTm-RJ+We52vfFiPjbiThGZGH0M3g@mail.gmail.com>

Actually no. I'm writing a script to control FS, incoming calls will be
screened and routing will de done via this script... I already have this in
Lua, but everyting a call comes in a new script is lunched, and I don't
want that anymore. So I'm trying to write an ESL script wich will connect
to FS via ESL and receive events, act accordingly and instruct fs what to
do, in terms of routing, channel variables definition (for later use in
xml_curl_cdr, etc.

Maybe my approach is not the correct one...

Thanks for your help...

David

On Thu, Aug 2, 2012 at 9:07 PM, Michael Collins <msc at freeswitch.org> wrote:

> Well, it all depends on how you're calling this script. Is it being called
> from the dialplan?
> -MC
>
>
> On Thu, Aug 2, 2012 at 11:25 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Ouch!
>>
>> Ok, this is not like in Lua... Ok, I'll try that.... is there any
>> documentation regarding what can and can't be set and the conditions for
>> setting them? What about bridging channels, etc?
>>
>> :(
>>
>> David
>>
>> On Thu, Aug 2, 2012 at 8:19 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Can't use execute and set unless this is an outbound ESL connection,
>>> which it seems not to be. I'd try this:
>>>
>>> $con->api("uuid_setvar","$uuid customer_company $local_variable");
>>>
>>> -MC
>>>
>>>
>>> On Thu, Aug 2, 2012 at 11:01 AM, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> Hello Michael,
>>>>
>>>> Yes, there's a call in progress. Executing it like you said worked.
>>>> (api)
>>>>
>>>>
>>>> Now, I'm trying to set a variable like so:
>>>>
>>>> $con->execute("set","customer_company             = " .
>>>>  $local_variable , $uuid );
>>>>
>>>> But it doesn't get set...
>>>>
>>>> :(
>>>>
>>>> Thanks for your help!
>>>>
>>>> David
>>>>
>>>>
>>>> On Thu, Aug 2, 2012 at 6:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> David,
>>>>>
>>>>> Do you have a call in progress at this point? If not then you'll need
>>>>> to supply a uuid of a live call, as mentioned here:
>>>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute
>>>>>
>>>>> Remember, "execute" means "execute dialplan application" so if there's
>>>>> no channel then "execute" really doesn't mean a whole lot.
>>>>>
>>>>> Alternatively you could try something like this:
>>>>>
>>>>> $con->api("log","WARNING Don't cross the streams!!");
>>>>>
>>>>> Remember this rule of thumb: pretty much anything you type at fs_cli
>>>>> is an "API" and therefore you can use $con->api(), whereas anything that is
>>>>> a diaplan application requires an actual channel on which to run
>>>>> $con->execute().
>>>>>
>>>>> Hope that makes sense... :)
>>>>>
>>>>> -Michael
>>>>>
>>>>> On Thu, Aug 2, 2012 at 8:43 AM, David Villasmil <
>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>
>>>>>> Hello Guys,
>>>>>>
>>>>>> I'm starting off with ESL, which is cool, but I'm trying to log to
>>>>>> the console like Lua's "freeswitch.consoleLog("info", "BLAH BLAH");" doing
>>>>>> it like:
>>>>>>
>>>>>> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>>>>>> $con->execute("log", "1, BlahBlah");
>>>>>>
>>>>>>
>>>>>> But nothing gets in the log files or console... and I can't find any
>>>>>> documentation as to how to log using "execute"...
>>>>>>
>>>>>> any ideas?
>>>>>>
>>>>>> Thanks!
>>>>>>
>>>>>> David
>>>>>>
>>>>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sazzadbinkamal at gmail.com  Thu Aug  2 23:26:54 2012
From: sazzadbinkamal at gmail.com (Sazzad)
Date: Fri, 3 Aug 2012 01:26:54 +0600
Subject: [Freeswitch-users] FreeSwitch Error: "username param is
	REQUIRED!"
In-Reply-To: <CADyf1QM87w4hQByq8nz2ujeH_z9a5+R43aJmG4Prmpg=M3d4ag@mail.gmail.com>
References: <CADyf1QM87w4hQByq8nz2ujeH_z9a5+R43aJmG4Prmpg=M3d4ag@mail.gmail.com>
Message-ID: <CADyf1QPgeQ6MBtxhSGv1ATkTRBvmOGUQy9w0aRnOyQtMeTXsVg@mail.gmail.com>

Hi Michael,

Thank you very much for your reply. I've tried out that as well. Same
result. Just one simple question. Can you tell me whether this error
message is an indication of failure? Or I can safely ignore it? I'm
currently testing on my local machine after I tried (and failed) on a live
server.

-- 
Sincerely,
Sazzad Bin Kamal
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From msc at freeswitch.org  Thu Aug  2 23:42:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 2 Aug 2012 12:42:27 -0700
Subject: [Freeswitch-users] FreeSwitch Error: "username param is
	REQUIRED!"
In-Reply-To: <CADyf1QPgeQ6MBtxhSGv1ATkTRBvmOGUQy9w0aRnOyQtMeTXsVg@mail.gmail.com>
References: <CADyf1QM87w4hQByq8nz2ujeH_z9a5+R43aJmG4Prmpg=M3d4ag@mail.gmail.com>
	<CADyf1QPgeQ6MBtxhSGv1ATkTRBvmOGUQy9w0aRnOyQtMeTXsVg@mail.gmail.com>
Message-ID: <CAKzWOxWcJ0enk0ELroC9s6pBdm4sdz4vtmHf3UH1aRs9nXsNyQ@mail.gmail.com>

If you're still getting the error after putting in a password then
something isn't configured correctly. Paste your entire gateway xml file to
pastebin.freeswitch.org and the gang here will offer you some suggestions.

-MC

On Thu, Aug 2, 2012 at 12:26 PM, Sazzad <sazzadbinkamal at gmail.com> wrote:

> Hi Michael,
>
> Thank you very much for your reply. I've tried out that as well. Same
> result. Just one simple question. Can you tell me whether this error
> message is an indication of failure? Or I can safely ignore it? I'm
> currently testing on my local machine after I tried (and failed) on a live
> server.
>
> --
> Sincerely,
> Sazzad Bin Kamal
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From sazzadbinkamal at gmail.com  Thu Aug  2 23:59:25 2012
From: sazzadbinkamal at gmail.com (Sazzad)
Date: Fri, 3 Aug 2012 01:59:25 +0600
Subject: [Freeswitch-users] FreeSwitch Error: "username param is
	REQUIRED!"
In-Reply-To: <CAKzWOxWcJ0enk0ELroC9s6pBdm4sdz4vtmHf3UH1aRs9nXsNyQ@mail.gmail.com>
References: <CADyf1QM87w4hQByq8nz2ujeH_z9a5+R43aJmG4Prmpg=M3d4ag@mail.gmail.com>
	<CADyf1QPgeQ6MBtxhSGv1ATkTRBvmOGUQy9w0aRnOyQtMeTXsVg@mail.gmail.com>
	<CAKzWOxWcJ0enk0ELroC9s6pBdm4sdz4vtmHf3UH1aRs9nXsNyQ@mail.gmail.com>
Message-ID: <CADyf1QOht3-kVdSAOo+1UC2RKwCd0bSA_gQszErxee+Gxvg_dQ@mail.gmail.com>

On Fri, Aug 3, 2012 at 1:42 AM, Michael Collins <msc at freeswitch.org> wrote:

> If you're still getting the error after putting in a password then
> something isn't configured correctly. Paste your entire gateway xml file to
> pastebin.freeswitch.org and the gang here will offer you some suggestions.


You guys rock. :D

Actually the file <http://pastebin.freeswitch.org/19630> is
pretty bare-bone. I'm trying to use Alcazar
Networks<http://www.alcazarnetworks.com/>.
I get the same error at my server which is *registered* with Alcazar and on
my local machine which is not registered at all.

-- 
Sincerely,
Sazzad Bin Kamal
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From sazzadbinkamal at gmail.com  Fri Aug  3 00:14:08 2012
From: sazzadbinkamal at gmail.com (Sazzad)
Date: Fri, 3 Aug 2012 02:14:08 +0600
Subject: [Freeswitch-users] FreeSwitch Error: "username param is
	REQUIRED!"
In-Reply-To: <CADyf1QOht3-kVdSAOo+1UC2RKwCd0bSA_gQszErxee+Gxvg_dQ@mail.gmail.com>
References: <CADyf1QM87w4hQByq8nz2ujeH_z9a5+R43aJmG4Prmpg=M3d4ag@mail.gmail.com>
	<CADyf1QPgeQ6MBtxhSGv1ATkTRBvmOGUQy9w0aRnOyQtMeTXsVg@mail.gmail.com>
	<CAKzWOxWcJ0enk0ELroC9s6pBdm4sdz4vtmHf3UH1aRs9nXsNyQ@mail.gmail.com>
	<CADyf1QOht3-kVdSAOo+1UC2RKwCd0bSA_gQszErxee+Gxvg_dQ@mail.gmail.com>
Message-ID: <CADyf1QOVkVp-qq5sU8inj3Zur+C82AFPzUvQQdxiKEoeESobAw@mail.gmail.com>

Hi,

The people at there told not to use authentication but to use bridge
command. I'm going to try it. Will dig the issue later. :D

-- 
Sincerely,
Sazzad Bin Kamal
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From paul at cupis.co.uk  Fri Aug  3 00:23:42 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Thu, 02 Aug 2012 21:23:42 +0100
Subject: [Freeswitch-users] FreeSwitch Error: "username param is
	REQUIRED!"
In-Reply-To: <CADyf1QOht3-kVdSAOo+1UC2RKwCd0bSA_gQszErxee+Gxvg_dQ@mail.gmail.com>
References: <CADyf1QM87w4hQByq8nz2ujeH_z9a5+R43aJmG4Prmpg=M3d4ag@mail.gmail.com>
	<CADyf1QPgeQ6MBtxhSGv1ATkTRBvmOGUQy9w0aRnOyQtMeTXsVg@mail.gmail.com>
	<CAKzWOxWcJ0enk0ELroC9s6pBdm4sdz4vtmHf3UH1aRs9nXsNyQ@mail.gmail.com>
	<CADyf1QOht3-kVdSAOo+1UC2RKwCd0bSA_gQszErxee+Gxvg_dQ@mail.gmail.com>
Message-ID: <501AE1CE.9080702@cupis.co.uk>

On 02/08/12 20:59, Sazzad wrote:
> Actually the file <http://pastebin.freeswitch.org/19630> is
> pretty bare-bone. I'm trying to use Alcazar Networks
> <http://www.alcazarnetworks.com/>. I get the same error at my server
> which is /registered/ with Alcazar and on my local machine which is not
> registered at all.

You don't have a closing </gateway> tag in that example. Not sure if 
that is your actual issue though.

Have you reloaded the configuration/rescaned the sofia gateways since 
changing the file? Could you give us an example of the console log 
showing the fault, please?

Regards,


From sazzadbinkamal at gmail.com  Fri Aug  3 00:32:50 2012
From: sazzadbinkamal at gmail.com (Sazzad)
Date: Fri, 3 Aug 2012 02:32:50 +0600
Subject: [Freeswitch-users] FreeSwitch Error: "username param is
	REQUIRED!"
In-Reply-To: <501AE1CE.9080702@cupis.co.uk>
References: <CADyf1QM87w4hQByq8nz2ujeH_z9a5+R43aJmG4Prmpg=M3d4ag@mail.gmail.com>
	<CADyf1QPgeQ6MBtxhSGv1ATkTRBvmOGUQy9w0aRnOyQtMeTXsVg@mail.gmail.com>
	<CAKzWOxWcJ0enk0ELroC9s6pBdm4sdz4vtmHf3UH1aRs9nXsNyQ@mail.gmail.com>
	<CADyf1QOht3-kVdSAOo+1UC2RKwCd0bSA_gQszErxee+Gxvg_dQ@mail.gmail.com>
	<501AE1CE.9080702@cupis.co.uk>
Message-ID: <CADyf1QO-=SkOQKnrjMFUYeTj5+C4MSKHMhExKBW4_p7mvs3ECw@mail.gmail.com>

>
> You don't have a closing </gateway> tag in that example. Not sure if
> that is your actual issue though.
>

Umm, well that might be missing there, its not causing the problem. I've
tried several approaches before posting. The only left is to debug. Like
printing that "profile.username" string and if its really null or where did
it get lost etc.

Have you reloaded the configuration/rescaned the sofia gateways since
> changing the file? Could you give us an example of the console log
> showing the fault, please?
>

Yeah did that. This is the exact copy-paste of the relevant output:

2012-08-03 01:02:19.768103 [NOTICE] sofia.c:2168 Waiting for worker thread
> 2012-08-03 01:02:19.768103 [NOTICE] sofia_glue.c:5574 deleted gateway
> example.com from profile external
> 2012-08-03 01:02:19.768103 [NOTICE] sofia.c:4782 Started Profile external
> [sofia_reg_external]
> 2012-08-03 01:02:19.788155 [ERR] sofia.c:2606 ERROR: username param is
> REQUIRED!
> 2012-08-03 01:02:19.788155 [NOTICE] sofia_reg.c:2904 Added gateway '
> example.com' to profile 'external'


-- 
Sincerely,
Sazzad Bin Kamal
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From paul at cupis.co.uk  Fri Aug  3 01:01:17 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Thu, 02 Aug 2012 22:01:17 +0100
Subject: [Freeswitch-users] FreeSwitch Error: "username param is
	REQUIRED!"
In-Reply-To: <CADyf1QO-=SkOQKnrjMFUYeTj5+C4MSKHMhExKBW4_p7mvs3ECw@mail.gmail.com>
References: <CADyf1QM87w4hQByq8nz2ujeH_z9a5+R43aJmG4Prmpg=M3d4ag@mail.gmail.com>
	<CADyf1QPgeQ6MBtxhSGv1ATkTRBvmOGUQy9w0aRnOyQtMeTXsVg@mail.gmail.com>
	<CAKzWOxWcJ0enk0ELroC9s6pBdm4sdz4vtmHf3UH1aRs9nXsNyQ@mail.gmail.com>
	<CADyf1QOht3-kVdSAOo+1UC2RKwCd0bSA_gQszErxee+Gxvg_dQ@mail.gmail.com>
	<501AE1CE.9080702@cupis.co.uk>
	<CADyf1QO-=SkOQKnrjMFUYeTj5+C4MSKHMhExKBW4_p7mvs3ECw@mail.gmail.com>
Message-ID: <501AEA9D.1030507@cupis.co.uk>

On 02/08/12 21:32, Sazzad wrote:
> This is the exact copy-paste of the relevant output:
>
>     2012-08-03 01:02:19.768103 [NOTICE] sofia.c:2168 Waiting for worker
>     thread
>     2012-08-03 01:02:19.768103 [NOTICE] sofia_glue.c:5574 deleted
>     gateway example.com <http://example.com> from profile external
>     2012-08-03 01:02:19.768103 [NOTICE] sofia.c:4782 Started Profile
>     external [sofia_reg_external]
>     2012-08-03 01:02:19.788155 [ERR] sofia.c:2606 ERROR: username param
>     is REQUIRED!
>     2012-08-03 01:02:19.788155 [NOTICE] sofia_reg.c:2904 Added gateway
>     'example.com <http://example.com>' to profile 'external'

This is odd then. I don't know if the issue is occuring where you think 
it is. Do you have any other gateways setup in profile external? Can you 
post the output of 'sofia status' with minimal changes, please?

For reference, the following definitely works for a provider which does 
not require username/password (i.e. IP authentication only), I've only 
obfuscated the actual IP address from the realm in this example:

   <include>
     <gateway name="provider">
       <param name="username" value="not-used"/>
       <param name="password" value="not-used"/>
       <param name="realm" value="192.0.2.1"/>
       <param name="register" value="false"/>
       <param name="caller-id-in-from" value="true"/>
     </gateway>
   </include>

Regards,



From sazzadbinkamal at gmail.com  Fri Aug  3 01:22:02 2012
From: sazzadbinkamal at gmail.com (Sazzad)
Date: Fri, 3 Aug 2012 03:22:02 +0600
Subject: [Freeswitch-users] FreeSwitch Error: "username param is
	REQUIRED!"
In-Reply-To: <501AEA9D.1030507@cupis.co.uk>
References: <CADyf1QM87w4hQByq8nz2ujeH_z9a5+R43aJmG4Prmpg=M3d4ag@mail.gmail.com>
	<CADyf1QPgeQ6MBtxhSGv1ATkTRBvmOGUQy9w0aRnOyQtMeTXsVg@mail.gmail.com>
	<CAKzWOxWcJ0enk0ELroC9s6pBdm4sdz4vtmHf3UH1aRs9nXsNyQ@mail.gmail.com>
	<CADyf1QOht3-kVdSAOo+1UC2RKwCd0bSA_gQszErxee+Gxvg_dQ@mail.gmail.com>
	<501AE1CE.9080702@cupis.co.uk>
	<CADyf1QO-=SkOQKnrjMFUYeTj5+C4MSKHMhExKBW4_p7mvs3ECw@mail.gmail.com>
	<501AEA9D.1030507@cupis.co.uk>
Message-ID: <CADyf1QPvdpC7RJnNkUQ_SbvRbhAhSUCAkTpTXob7SN56zdzS+w@mail.gmail.com>

>
> This is odd then.


(@_@)


> I don't know if the issue is occuring where you think
> it is. Do you have any other gateways setup in profile external?


Nope!


> Can you post the output of 'sofia status' with minimal changes, please?
>

Sure, I can. Though I don't think it'll be of much help to you.
Here<http://pastebin.freeswitch.org/19631>it is.


> For reference, the following definitely works for a provider which does
> not require username/password (i.e. IP authentication only), I've only
> obfuscated the actual IP address from the realm in this example:
>
>    <include>
>      <gateway name="provider">
>        <param name="username" value="not-used"/>
>        <param name="password" value="not-used"/>
>        <param name="realm" value="192.0.2.1"/>
>        <param name="register" value="false"/>
>        <param name="caller-id-in-from" value="true"/>
>      </gateway>
>    </include>


I see. Hmm, but Alcazar support adviced to use Freeswitch bridge command to
point to ip. Probably it is same as Asterisk bridge()?! And it definitely
is not seeming like a cli command rather an application to be used in
dialplan, I guess. By the way, I registered Alcazar for outbound
termination.

May be this context may help you find out some reason:

I'm trying to use Newfies <http://newfies-dialer.org> dialer.
Plivo<http://plivo.com>has been installed on the server, so we'll not
be using their API (or
sending request to their cloud server).

After a fresh install I simply did the following according to Newfies doc:

1. copy-paste the sip_profile/external/example.com to alcazarnetworks.com
2. change relevant fields, EXACTLY same as you provided in the above
example. EXCEPT "realm" and "caller-id-in-form" fields. I used real
username and password provided by Alcazar by the way.
3. Run command : "sip profile external restart reloadxml" and got the error!
4. So I tried it on my local machine (which is not registered with Alcazar)
and got the same error!

That's all I can tell.

-- 
Sincerely,
Sazzad Bin Kamal
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From all.eforums at gmail.com  Fri Aug  3 02:58:33 2012
From: all.eforums at gmail.com (A E G)
Date: Thu, 2 Aug 2012 18:58:33 -0400
Subject: [Freeswitch-users] FreeSwitch Error: "username param is
	REQUIRED!"
In-Reply-To: <CADyf1QPvdpC7RJnNkUQ_SbvRbhAhSUCAkTpTXob7SN56zdzS+w@mail.gmail.com>
References: <CADyf1QM87w4hQByq8nz2ujeH_z9a5+R43aJmG4Prmpg=M3d4ag@mail.gmail.com>
	<CADyf1QPgeQ6MBtxhSGv1ATkTRBvmOGUQy9w0aRnOyQtMeTXsVg@mail.gmail.com>
	<CAKzWOxWcJ0enk0ELroC9s6pBdm4sdz4vtmHf3UH1aRs9nXsNyQ@mail.gmail.com>
	<CADyf1QOht3-kVdSAOo+1UC2RKwCd0bSA_gQszErxee+Gxvg_dQ@mail.gmail.com>
	<501AE1CE.9080702@cupis.co.uk>
	<CADyf1QO-=SkOQKnrjMFUYeTj5+C4MSKHMhExKBW4_p7mvs3ECw@mail.gmail.com>
	<501AEA9D.1030507@cupis.co.uk>
	<CADyf1QPvdpC7RJnNkUQ_SbvRbhAhSUCAkTpTXob7SN56zdzS+w@mail.gmail.com>
Message-ID: <CADzY+JhVey2Jgukgp3ZsPJejZTNjq=eCBWObNC_PeVSZaWsjLg@mail.gmail.com>

I don't see anywhere where this " alcazarnetworks-outbound " gateway is
even being loaded. Nor do I see where FS would figure out how to reach this
alcaz-outbound realm. Are these being looked up from DNS? If not, then how
do you propose sofia finds the IP to contact this gateway on? I think with
the bridge command you might have to still tell it the IP. If you tell it
the IP, then why is the definition of the gateway needed if you're not
registering.

Haha sorry I'm not all that of an expert at FS either so maybe someone more
knowledgeable can respond to this and help us both.

I think what you need to run, as mentioned previously is

fs at internal>sofia profile external gwlist up <-- this should show you the
gateways that are loaded and ready (I think)

if you don't see your  alcazarnetworks-outbound  gateway there, try running

fs at internal>sofia profile external rescan reloadxml <-- this should rescan
your gateways in the directory and you should see alcazarnetworks-outbound
load up OR it should give you some error as to why it can't load it up.
Also, don't forget to include the </gateway> in the config file. I'm
surprised that this doesn't show up as an error in the console when you
load it up.

HTH
aeg

On Thu, Aug 2, 2012 at 5:22 PM, Sazzad <sazzadbinkamal at gmail.com> wrote:

> This is odd then.
>
>
> (@_@)
>
>
>> I don't know if the issue is occuring where you think
>> it is. Do you have any other gateways setup in profile external?
>
>
> Nope!
>
>
>> Can you post the output of 'sofia status' with minimal changes, please?
>>
>
> Sure, I can. Though I don't think it'll be of much help to you. Here<http://pastebin.freeswitch.org/19631>it is.
>
>
>> For reference, the following definitely works for a provider which does
>> not require username/password (i.e. IP authentication only), I've only
>> obfuscated the actual IP address from the realm in this example:
>>
>>    <include>
>>      <gateway name="provider">
>>        <param name="username" value="not-used"/>
>>        <param name="password" value="not-used"/>
>>        <param name="realm" value="192.0.2.1"/>
>>        <param name="register" value="false"/>
>>        <param name="caller-id-in-from" value="true"/>
>>      </gateway>
>>    </include>
>
>
> I see. Hmm, but Alcazar support adviced to use Freeswitch bridge command
> to point to ip. Probably it is same as Asterisk bridge()?! And it
> definitely is not seeming like a cli command rather an application to be
> used in dialplan, I guess. By the way, I registered Alcazar for outbound
> termination.
>
> May be this context may help you find out some reason:
>
> I'm trying to use Newfies <http://newfies-dialer.org> dialer. Plivo<http://plivo.com>has been installed on the server, so we'll not be using their API (or
> sending request to their cloud server).
>
> After a fresh install I simply did the following according to Newfies doc:
>
> 1. copy-paste the sip_profile/external/example.com to alcazarnetworks.com
> 2. change relevant fields, EXACTLY same as you provided in the above
> example. EXCEPT "realm" and "caller-id-in-form" fields. I used real
> username and password provided by Alcazar by the way.
> 3. Run command : "sip profile external restart reloadxml" and got the
> error!
> 4. So I tried it on my local machine (which is not registered with
> Alcazar) and got the same error!
>
> That's all I can tell.
>
> --
> Sincerely,
> Sazzad Bin Kamal
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Fri Aug  3 03:23:33 2012
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 2 Aug 2012 20:23:33 -0300
Subject: [Freeswitch-users] BDA and Voicemail main
Message-ID: <CABH8QX3V9aheF6TOb3_X0iSyv6F4nMJe3Sqvw229gHpbbc-e6w@mail.gmail.com>

Guys, is it expected that voicemail main menu would not allow overlap
dialing when the calling leg has bind digit action set on the channel?

If it is not expected, I would open a Jira, but I just wanted to make sure
this is not expected.

Regards,
Jo?o Mesquita
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From all.eforums at gmail.com  Fri Aug  3 08:01:30 2012
From: all.eforums at gmail.com (A E G)
Date: Fri, 3 Aug 2012 00:01:30 -0400
Subject: [Freeswitch-users] Error: 88 Incompatible Destination after upgrade
Message-ID: <CADzY+Jh6+UkESj=W=Djjj6PcGqEcR7b2BsUWo_TW2ax+gd5U8g@mail.gmail.com>

Gents,

running version: 1.2.0-rc2+git~20120731T213556Z~e97da8e20a

have been fighting with this for an hour or more...have done a bit of
research on the list and Google itself, scoured the Wiki etc. but can't
seem to figure out where to set the codecs to be "well recd" by the remote
SIP peer. This used to work fine until I upgraded 2 days ago and it stopped
working without my doing / changing anything.

The remote SIP peer is {*} which doesn't like multiple m= lines with
differing ptime values.

Debug on near-end says: "Originate Resulted in Error Cause: 88
[INCOMPATIBLE_DESTINATION]"

Full SDP here:

v=0
o=FreeSWITCH 1343934933 1343934934 IN IP4 192.168.1.80
s=FreeSWITCH
c=IN IP4 192.168.1.80
t=0 0
m=audio 28884 RTP/AVP 98 8 3 101 13
a=rtpmap:98 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=audio 28884 RTP/AVP 0 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:40
a=sendrecv

Far end says:  Rejecting non-primary audio stream: audio 28884 RTP/AVP 0
101 13

I have tried to play around with the codec globals in vars.xml, to no avail.

have also added stuff directly in the dialplan like so:

<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="absolute_codec_string=PCMU at 40i"/>
<action application="set" data="sdp_m_per_ptime=false"/>
<action application="bridge" data="sofia/gateway/mycore1/1111"/>


but no dice.

Tried to remove the "absolute_codec_string", still no dice.

I read about the "sdp_m_per_ptime"  that lets me "cheat" but that ain't
doing anything either.

What gives?

Thx in advance
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From krice at freeswitch.org  Fri Aug  3 09:05:43 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 03 Aug 2012 00:05:43 -0500
Subject: [Freeswitch-users] FreeSwitch Error: "username param is
 REQUIRED!"
In-Reply-To: <CADzY+JhVey2Jgukgp3ZsPJejZTNjq=eCBWObNC_PeVSZaWsjLg@mail.gmail.com>
Message-ID: <CC40C657.3ACEC%krice@freeswitch.org>

Just put nothing or BS in the username and password fields for the gateway
if you are using IP auth on the other side like with alcazar...

If they don?t 407 the gateway doesn?t mind having the extra info
there...also keep in mind to set the caller id in from field setting on the
gateway


On 8/2/12 5:58 PM, "A E G" <all.eforums at gmail.com> wrote:

> I don't see anywhere where this " alcazarnetworks-outbound?" gateway is even
> being loaded. Nor do I see where FS would figure out how to reach this
> alcaz-outbound realm. Are these being looked up from DNS? If not, then how do
> you propose sofia finds the IP to contact this gateway on? I think with the
> bridge command you might have to still tell it the IP. If you tell it the IP,
> then why is the definition of the gateway needed if you're not registering.?
> 
> Haha sorry I'm not all that of an expert at FS either so maybe someone more
> knowledgeable can respond to this and help us both.?
> 
> I think what you need to run, as mentioned previously is
> 
> fs at internal>sofia profile external gwlist up <-- this should show you the
> gateways that are loaded and ready (I think)
> 
> if you don't see your? alcazarnetworks-outbound? gateway there, try running
> 
> fs at internal>sofia profile external rescan reloadxml <-- this should rescan
> your gateways in the directory and you should see alcazarnetworks-outbound
> load up OR it should give you some error as to why it can't load it up. Also,
> don't forget to include the </gateway> in the config file. I'm surprised that
> this doesn't show up as an error in the console when you load it up.?
> 
> HTH
> aeg
> 
> On Thu, Aug 2, 2012 at 5:22 PM, Sazzad <sazzadbinkamal at gmail.com> wrote:
>>> This is odd then.
>> 
>> (@_@)
>> ?
>>> I don't know if the issue is occuring where you think
>>> it is. Do you have any other gateways setup in profile external?
>> 
>> Nope!
>> ?
>>>  Can you?post the output of 'sofia status' with minimal changes, please?
>> 
>> Sure, I can. Though I don't think it'll be of much help to you. Here
>> <http://pastebin.freeswitch.org/19631>  it is.
>> ?
>>> For reference, the following definitely works for a provider which does
>>> not require username/password (i.e. IP authentication only), I've only
>>> obfuscated the actual IP address from the realm in this example:
>>> 
>>> ? ?<include>
>>> ? ? ?<gateway name="provider">
>>> ? ? ? ?<param name="username" value="not-used"/>
>>> ? ? ? ?<param name="password" value="not-used"/>
>>> ? ? ? ?<param name="realm" value="192.0.2.1"/>
>>> ? ? ? ?<param name="register" value="false"/>
>>> ? ? ? ?<param name="caller-id-in-from" value="true"/>
>>> ? ? ?</gateway>
>>> ? ?</include>
>> 
>> I see. Hmm, but Alcazar support adviced to use Freeswitch bridge command to
>> point to ip. Probably it is same as Asterisk bridge()?! And it definitely is
>> not seeming like a cli command rather an application to be used in dialplan,
>> I guess. By the way, I registered Alcazar for outbound termination.
>> 
>> May be this context may help you find out some reason:
>> 
>> I'm trying to use Newfies <http://newfies-dialer.org>  dialer. Plivo
>> <http://plivo.com>  has been installed on the server, so we'll not be using
>> their API (or sending request to their cloud server).
>> 
>> After a fresh install I simply did the following according to Newfies doc:
>> 
>> 1. copy-paste the sip_profile/external/example.com <http://example.com>  to
>> alcazarnetworks.com <http://alcazarnetworks.com>
>> 2. change relevant fields, EXACTLY same as you provided in the above example.
>> EXCEPT "realm" and "caller-id-in-form" fields. I used real username and
>> password provided by Alcazar by the way.
>> 3. Run command : "sip profile external restart reloadxml" and got the error!
>> 4. So I tried it on my local machine (which is not registered with Alcazar)
>> and got the same error!
>> 
>> That's all I can tell.

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From all.eforums at gmail.com  Fri Aug  3 09:44:44 2012
From: all.eforums at gmail.com (A E G)
Date: Fri, 3 Aug 2012 01:44:44 -0400
Subject: [Freeswitch-users] SOLVED: Asterisk don't like multiple m= lines in
 the SDP [Was: Error: 88 Incompatible Destination after upgrade]
Message-ID: <CADzY+JiUnUY+U=jrnVzBWNgYqp42jnEksHGA-DWfMnbt-Use9A@mail.gmail.com>

Well turns out learning to use Google better always helps.

Found a nugget of wisdom that <action application="set"
data="sdp_m_per_ptime=false"/> should've actually been in the vars.xml

Seems to have fixed it.



On Fri, Aug 3, 2012 at 12:01 AM, A E G <all.eforums at gmail.com> wrote:

> Gents,
>
> running version: 1.2.0-rc2+git~20120731T213556Z~e97da8e20a
>
> have been fighting with this for an hour or more...have done a bit of
> research on the list and Google itself, scoured the Wiki etc. but can't
> seem to figure out where to set the codecs to be "well recd" by the remote
> SIP peer. This used to work fine until I upgraded 2 days ago and it stopped
> working without my doing / changing anything.
>
> The remote SIP peer is {*} which doesn't like multiple m= lines with
> differing ptime values.
>
> Debug on near-end says: "Originate Resulted in Error Cause: 88
> [INCOMPATIBLE_DESTINATION]"
>
> Full SDP here:
>
> v=0
> o=FreeSWITCH 1343934933 1343934934 IN IP4 192.168.1.80
> s=FreeSWITCH
> c=IN IP4 192.168.1.80
> t=0 0
> m=audio 28884 RTP/AVP 98 8 3 101 13
> a=rtpmap:98 L16/16000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> m=audio 28884 RTP/AVP 0 101 13
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:40
> a=sendrecv
>
> Far end says:  Rejecting non-primary audio stream: audio 28884 RTP/AVP 0
> 101 13
>
> I have tried to play around with the codec globals in vars.xml, to no
> avail.
>
> have also added stuff directly in the dialplan like so:
>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set" data="absolute_codec_string=PCMU at 40i"/>
> <action application="set" data="sdp_m_per_ptime=false"/>
> <action application="bridge" data="sofia/gateway/mycore1/1111"/>
>
>
> but no dice.
>
> Tried to remove the "absolute_codec_string", still no dice.
>
> I read about the "sdp_m_per_ptime"  that lets me "cheat" but that ain't
> doing anything either.
>
> What gives?
>
> Thx in advance
>
>
>
>
>
>
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From hkalyoncu at gmail.com  Fri Aug  3 10:41:36 2012
From: hkalyoncu at gmail.com (huseyin kalyoncu)
Date: Fri, 3 Aug 2012 09:41:36 +0300
Subject: [Freeswitch-users] freeswitch_licence_server problem
In-Reply-To: <CAB+ia=2DkqDXvPZEvRTWqVYF=1i3S4ViryEbwiJM9FxrtM9=Wg@mail.gmail.com>
References: <CAB+ia=2DkqDXvPZEvRTWqVYF=1i3S4ViryEbwiJM9FxrtM9=Wg@mail.gmail.com>
Message-ID: <CAPf+86UZZT2h87frz=gC0hfJbBq7Qo7Rf_AjbMP_5q79NXq9XA@mail.gmail.com>

i have the same problem. but i cant get any response from
consulting at freeswitch.org

anybody solved this kind of problem before?

thanks
huseyin

On Wed, Aug 1, 2012 at 2:40 PM, Ale <ale975 at gmail.com> wrote:

> Hello,
>
> short: after installation fs show "Can't contact licence server." and
> freeswitch_licence_server in console show "Unrecognised resource
> G.729A/0".
>
> I've purchased a license, downloaded 194 installer, mod_com_g729.so is
> in lib64/freeswitch/mod dir, and license file without licenze.zip is
> in /etc/freeswitch.
> Freeswitch run as user freeswitch user on a centos.
> Where i unload mod_g729 and load the mod_com, server correctly starts.
> I also try to kill the server,
> and manually start it a root user, but nothing change.
>
> could someone give me any hints?
> Thanks alessandro
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From admin at blindi.net  Fri Aug  3 11:16:26 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Fri, 3 Aug 2012 09:16:26 +0200 (CEST)
Subject: [Freeswitch-users] Question how can i get variables in lua from
 xml dialplan?
In-Reply-To: <E5C097497D1549998A432240B0C497E5@gmail.com>
References: <CAE2A9TCfrCw1=9ScVzsCcnP0CvaYVpMbHPAm19ChwY3rEQrN8A@mail.gmail.com>
	<Pine.LNX.4.64.1207272147390.27709@1a-8140.antagus.de>
	<Pine.LNX.4.64.1207291440510.15832@1a-8140.antagus.de>
	<CAAdTMa_9G1hVicxk_gG5xyTHZpMRk+3tP=b2Ceaj1UZ9H9Oh7A@mail.gmail.com>
	<CANBhyPBzLU-ARsZon9ZwhHJk0SvBBYP2TFPxJV6F1iDM1kr=Sw@mail.gmail.com>
	<Pine.LNX.4.64.1207300907590.22790@1a-8140.antagus.de>
	<E5C097497D1549998A432240B0C497E5@gmail.com>
Message-ID: <Pine.LNX.4.64.1208030915010.25711@1a-8140.antagus.de>

Hi Seven, thanks for you nice help.
I have set the variables from the xml dialplan.

tnaks!


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From admin at blindi.net  Fri Aug  3 11:30:29 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Fri, 3 Aug 2012 09:30:29 +0200 (CEST)
Subject: [Freeswitch-users] Question how can i get variables in lua from
 xml dialplan?
In-Reply-To: <CAKzWOxXT6YibnTXCch6TQzRZrTriOFqTcpOTapD9iYUgQ4AdUA@mail.gmail.com>
References: <CAE2A9TCfrCw1=9ScVzsCcnP0CvaYVpMbHPAm19ChwY3rEQrN8A@mail.gmail.com>
	<Pine.LNX.4.64.1207272147390.27709@1a-8140.antagus.de>
	<Pine.LNX.4.64.1207291440510.15832@1a-8140.antagus.de>
	<CAAdTMa_9G1hVicxk_gG5xyTHZpMRk+3tP=b2Ceaj1UZ9H9Oh7A@mail.gmail.com>
	<CANBhyPBzLU-ARsZon9ZwhHJk0SvBBYP2TFPxJV6F1iDM1kr=Sw@mail.gmail.com>
	<Pine.LNX.4.64.1207300907590.22790@1a-8140.antagus.de>
	<CAKzWOxXT6YibnTXCch6TQzRZrTriOFqTcpOTapD9iYUgQ4AdUA@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1208030929320.25711@1a-8140.antagus.de>


Hi Michael, thanks for you nice help.
The script work perfect.


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From admin at blindi.net  Fri Aug  3 11:43:16 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Fri, 3 Aug 2012 09:43:16 +0200 (CEST)
Subject: [Freeswitch-users] Lua problem there are only two arguments
 evaluated
In-Reply-To: <CAJUJwthUm2fuxDet3SfH0tptnLDvB493Nc+NoQN9pZx9ecm+OQ@mail.gmail.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<Pine.LNX.4.64.1207311049050.2510@1a-8140.antagus.de>
	<CAJUJwtje2c3i_w9w0aqU0Hwmb0AqifuWS0gMqdAkrsrStBc1jQ@mail.gmail.com>
	<Pine.LNX.4.64.1207311129320.2808@1a-8140.antagus.de>
	<CAJUJwthUm2fuxDet3SfH0tptnLDvB493Nc+NoQN9pZx9ecm+OQ@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1208030940330.25711@1a-8140.antagus.de>

Hi SamyGo:
> If you don't get an output on :
>>
>> msgtime=msg_${strftime(%**Y_%m_%d_%H_%M_%S)}_${mailbox}.**alaw
>
>
> ${strftime(%**Y_%m_%d_%H_%M_%S)}
>
> I think the function strftime don't like the "_"

Thanks, i have set the variable, and paste  in to script as arguments.




From govoiper at gmail.com  Fri Aug  3 12:41:39 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 3 Aug 2012 13:41:39 +0500
Subject: [Freeswitch-users] Nibble Billing deducted balance from account
 even call doesn't come to my mobile number
In-Reply-To: <501A6085.2020308@bestnet.kharkov.ua>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<50194496.10804@bestnet.kharkov.ua>
	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>
	<501A5AF4.9020008@bestnet.kharkov.ua>
	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>
	<501A6085.2020308@bestnet.kharkov.ua>
Message-ID: <CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>

Hi
Your link isn't working for me !!

On Thu, Aug 2, 2012 at 4:12 PM, Evgeniy Movlyan
<evgeniy at bestnet.kharkov.ua>wrote:

> http://pastebin.com/enEmLVJz
>
> 02.08.2012 13:58, SamyGo ?????:
> > Ok,
> > Paste the whole call console log and share via pastebin.
> > Maybe find some other thing breaking it.
> > BR
> > Sammy
> > On Aug 2, 2012 3:52 PM, "Evgeniy Movlyan"<evgeniy at bestnet.kharkov.ua>
> > wrote:
> >
> >> I changed my dialplan, but got the same message =(
> >>
> >> 02.08.2012 07:49, SamyGo ?????:
> >>> You might wanna change your dialplan to something like this:
> >>>
> >>> <action application="bridge" data="{
> >>> ignore_early_media=true
> >> }sofia/gateway/${default_gateway}/${dialed_number}"/>
> >>> That's How I;ve done it, no wonder I use LUA but the dial-string is
> >> exactly
> >>> as below.
> >>>
> >>> I think there is a difference if you do it like above, It will set the
> >>> ignore_early_media on the B-leg session and  not on A-leg.
> >>>
> >>> This a snippet from the FS-1.6 book: *page:186*
> >>>
> >>> "Curly brackets are used "globally" for the duration of a call. Take
> the
> >>> following example where we are bridging a call to Darren's cell phone,
> >>> 203-829-3150. We
> >>> only want to ring the phone for 20 seconds, to avoid hitting voicemail.
> >>>
> >>> <application action="bridge"
> >>> data="{call_timeout=20}sofia/my_provider/2038293150">
> >>>
> >>> *The variable in brackets is utilized on the newly setup channel,
> >>> sofia/my_provider/2038293150.*"
> >>>
> >>>
> >>> Regards,
> >>> Sammy
> >>>
> >>>
> >>> On Wed, Aug 1, 2012 at 8:00 PM, Evgeniy Movlyan
> >>> <evgeniy at bestnet.kharkov.ua>wrote:
> >>>
> >>>> It did not work for me, i got the same message.
> >>>> My outbond extension:
> >>>>
> >>>> <extension name="AUTH_rad">
> >>>>     <condition field="${toll_allow}" expression="international"/>
> >>>>      <condition field="${AUTH_RESULT_RADIUS}" expression="^NOK$"
> >>>> break="on-true">
> >>>>       <action application="lua" data="callfailed.lua
> >> ${caller_id_number}"/>
> >>>>      </condition>
> >>>>      <condition field="${AUTH_RESULT_RADIUS}" expression="^OK$">
> >>>>       <action application="set"
> >>>> data="effective_caller_id_number=${outbound_caller_id_number}"/>
> >>>>       <action application="set"
> >>>> data="effective_caller_id_name=${outbound_caller_id_name}"/>
> >>>>       <action application="set" data="ignore_early_media=true"/>
> >>>>       <action application="bridge"
> >>>> data="sofia/gateway/${default_gateway}/${dialed_number}"/>
> >>>>      </condition>
> >>>> </extension>
> >>>>
> >>>>
> >>>> 31.07.2012 11:06, SamyGo ?????:
> >>>>> Hi,
> >>>>> Well its works perfect for me, do you guys have ignore_early_media
> set
> >> in
> >>>>> your outbound string, if no then set it and then see what happens.
> >>>>> On Jul 31, 2012 12:50 PM, "Evgeniy Movlyan"<
> evgeniy at bestnet.kharkov.ua
> >>>
> >>>>> wrote:
> >>>>>
> >>>>>> I have the same problem. When i am calling from one my extension to
> >>>>>> another all is ok, but when i am calling to external number i got
> this
> >>>>>> message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is not
> in
> >>>>>> answered state".
> >>>>>>
> >>>>>> 31.07.2012 10:35, virendra bhati ?????:
> >>>>>>> mod_nibblebill.c:465 Not billing
> >>>>>>> 97183008 - call is not in answered state
> >>>>>>
> >>>>>> --
> >>>>>> Evgeniy Movlyan,
> >>>>>> BestNet Ltd.
> >>>>>>
> >>>>>>
> >>>>
> >>
> _________________________________________________________________________
> >>>>>> Professional FreeSWITCH Consulting Services:
> >>>>>> consulting at freeswitch.org
> >>>>>> http://www.freeswitchsolutions.com
> >>>>>>
> >>>>>> 
> >>>>>> 
> >>>>>>
> >>>>>> Official FreeSWITCH Sites
> >>>>>> http://www.freeswitch.org
> >>>>>> http://wiki.freeswitch.org
> >>>>>> http://www.cluecon.com
> >>>>>>
> >>>>>> Join Us At ClueCon - Aug 7-9, 2012
> >>>>>>
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>> UNSUBSCRIBE:
> >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>> http://www.freeswitch.org
> >>>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>
> _________________________________________________________________________
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>>
> >>>>> 
> >>>>> 
> >>>>>
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://wiki.freeswitch.org
> >>>>> http://www.cluecon.com
> >>>>>
> >>>>> Join Us At ClueCon - Aug 7-9, 2012
> >>>>>
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>> --
> >>>> Evgeniy Movlyan,
> >>>> BestNet Ltd.
> >>>>
> >>>>
> >>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> Join Us At ClueCon - Aug 7-9, 2012
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> Join Us At ClueCon - Aug 7-9, 2012
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >> --
> >> Evgeniy Movlyan,
> >> BestNet Ltd.
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Evgeniy Movlyan,
> BestNet Ltd.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From govoiper at gmail.com  Fri Aug  3 12:52:07 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 3 Aug 2012 13:52:07 +0500
Subject: [Freeswitch-users] Lua problem there are only two arguments
	evaluated
In-Reply-To: <Pine.LNX.4.64.1208030940330.25711@1a-8140.antagus.de>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<Pine.LNX.4.64.1207311049050.2510@1a-8140.antagus.de>
	<CAJUJwtje2c3i_w9w0aqU0Hwmb0AqifuWS0gMqdAkrsrStBc1jQ@mail.gmail.com>
	<Pine.LNX.4.64.1207311129320.2808@1a-8140.antagus.de>
	<CAJUJwthUm2fuxDet3SfH0tptnLDvB493Nc+NoQN9pZx9ecm+OQ@mail.gmail.com>
	<Pine.LNX.4.64.1208030940330.25711@1a-8140.antagus.de>
Message-ID: <CAJUJwtimEQP_r2XMiuK7r5ncidHjM54nCBmdfH1Msa514K5n7A@mail.gmail.com>

So, you';ve changed the approach and set the required variable from XML
dialplan and now you dont need to set that as Argument !?

On Fri, Aug 3, 2012 at 12:43 PM, Thomas Hoellriegel <admin at blindi.net>wrote:

> Hi SamyGo:
> > If you don't get an output on :
> >>
> >> msgtime=msg_${strftime(%**Y_%m_%d_%H_%M_%S)}_${mailbox}.**alaw
> >
> >
> > ${strftime(%**Y_%m_%d_%H_%M_%S)}
> >
> > I think the function strftime don't like the "_"
>
> Thanks, i have set the variable, and paste  in to script as arguments.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From odermann at googlemail.com  Fri Aug  3 13:06:57 2012
From: odermann at googlemail.com (Dennis)
Date: Fri, 3 Aug 2012 11:06:57 +0200
Subject: [Freeswitch-users] Fax (RX): Missing name/header variable in
	socket!?
In-Reply-To: <CAKzWOxX-Doz18y+otDTGkPOumUOnA0=tD85tdrY8QS+e1pToiQ@mail.gmail.com>
References: <CALaOL+hYAxkVTEOLBCpCZB5H6LbZaLH_hW-UiagvP_XYAKOQbQ@mail.gmail.com>
	<CAKzWOxX-Doz18y+otDTGkPOumUOnA0=tD85tdrY8QS+e1pToiQ@mail.gmail.com>
Message-ID: <CALaOL+gfSKkp4VJ8W4-Z0MwZ3sxMYB9nyipFu7z8HpFrvTOPJA@mail.gmail.com>

here you can find the debug-log: http://pastebin.freeswitch.org/19632

kind regards
dennis


From B.Tietz at pinguin.ag  Fri Aug  3 13:09:24 2012
From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag)
Date: Fri, 3 Aug 2012 11:09:24 +0200
Subject: [Freeswitch-users] Remove Privacy: none Header in INVITE
Message-ID: <07BF4904977CC645B485E970424193AD118F3F7BBF@localhost>

Hi,

I have to send an INVITE with P-asserted-identity but without the Privacy: none header.
Right now I use

<action application="bridge" data="{sip_contact_user=12345,sip_cid_type=pid}sofia/gateway/gw1/..."/>

Is there anything I can set/unset to get this!?

regards,
Benjamin




From odermann at googlemail.com  Fri Aug  3 13:32:44 2012
From: odermann at googlemail.com (Dennis)
Date: Fri, 3 Aug 2012 11:32:44 +0200
Subject: [Freeswitch-users] Early media: Sending 180 instead of 183?
Message-ID: <CALaOL+gnBq-L8Nu3W76QJchzgYbw59BKGF44oWrzRbt0GDX7CA@mail.gmail.com>

hi,

we are doing a lot with early media for playing soundfiles/messages to
the caller without charging the caller, while they have to wait.

we are doing a "pre answer" -> fs sends a 183 message -> we start a
playback -> fs lets flow the media.

the problem is, that it happens sometimes, that the caller can not
hear anything. the problem might be, that in germany the specification
ITU-T Rec. Q.1912.5 does not necessarily map the 183 message to ISUP.

in germany it seems to be the best way, to start a playback with a 180
message instead of a 183 message.

is it possible to make fs sending a 180 message without making a
bridge (because we do not always want to connect to another target)?


thanks and kind regards
dennis


From Adam.Lappe at qsc.de  Fri Aug  3 13:53:25 2012
From: Adam.Lappe at qsc.de (Lappe, Adam)
Date: Fri, 3 Aug 2012 11:53:25 +0200
Subject: [Freeswitch-users] sendevent to Gateway
Message-ID: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EBF@CGNMSG01.oc.qsc.de>

Hi all,

i am trying to make the freeswitch send a event (or SIP Message) to my gateway.
I can only find examples how to send events to registered endpoint but not to my gateway.

Shouldn't it be possible to do this via the event socket?

Thanks in advance and best regards,
Adam
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From paul at cupis.co.uk  Fri Aug  3 14:24:13 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Fri, 3 Aug 2012 11:24:13 +0100
Subject: [Freeswitch-users] Remove Privacy: none Header in INVITE
In-Reply-To: <07BF4904977CC645B485E970424193AD118F3F7BBF@localhost>
References: <07BF4904977CC645B485E970424193AD118F3F7BBF@localhost>
Message-ID: <20120803102413.GA28145@eagle.cupis.co.uk>

On Fri, Aug 03, 2012 at 11:09:24AM +0200, B.Tietz at pinguin.ag wrote:
> I have to send an INVITE with P-asserted-identity but without the Privacy: none header.
> Right now I use
> 
> <action application="bridge" data="{sip_contact_user=12345,sip_cid_type=pid}sofia/gateway/gw1/..."/>
> 
> Is there anything I can set/unset to get this!?

This requirement from your carrier(?) seems a little odd, but you can do it
"the hard way" with something like this:

  <action application="set" data="sip_cid_type=none"/>
  <action application="unset" data="sip_h_Privacy"/>
  <action application="unset" data="sip_h_P-Asserted-Identity"/>
  <action application="set" data="sip_h_P-Asserted-Identity=&quot;NAME&quot; <sip:NUMBER at IP_ADDRESS>"/>

I don't know an easier way to just say "send P-A-ID, but not Privacy".

Regards,



From alex at thewinelake.com  Fri Aug  3 14:28:07 2012
From: alex at thewinelake.com (Alex)
Date: Fri, 03 Aug 2012 11:28:07 +0100
Subject: [Freeswitch-users] External profile & NAT
In-Reply-To: <CAKzWOxUW5ehDU+5cc=FCLnXX6=_iA0Gr0REwa0=wz9huztAExw@mail.gmail.com>
References: <4FDF35D1.306@the800group.com>
	<CAKzWOxUW5ehDU+5cc=FCLnXX6=_iA0Gr0REwa0=wz9huztAExw@mail.gmail.com>
Message-ID: <501BA7B7.1080709@thewinelake.com>

Does reload mod_sofia harm any calls in progress?
> After you add the external profile you need to reloadxml and then you 
> need to start the profile - assuming you didn't just restart 
> FreeSWITCH or reload mod_sofia.
>
> -MC
>
> On Mon, Jun 18, 2012 at 7:06 AM, ocset <ocset at the800group.com 
> <mailto:ocset at the800group.com>> wrote:
>
>     Hi
>
>     I would like to allow a person with a softphone app to register with
>     Freeswitch (extension 1111) when they are outside the office network.
>
>     I would like to to implement this using an external profile. According
>     to the wiki page (http://wiki.freeswitch.org/wiki/External_profile) I
>     just need to create a copy of the "conf/sip_profiles/external.xml"
>     file
>     and change the port number to 5090 (and also open the port on the
>     modem). When I do a "sofia status", this new gateway does not show up
>     and to be honest, I don't quite understand how that works and what
>     else
>     to read to make sense of what to do.
>
>     What are the ways in which other Freeswitch admins are granting access
>     to external devices?
>
>     ps. just to confirm, do I set external_rtp_ip and external_sip_ip
>     to the
>     fix IP address that my ISP has given me - the one that is shown by
>     www.whatsmyip.org <http://www.whatsmyip.org>?
>
>     Please help
>     Thanks
>     O.
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     FreeSWITCH-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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2177 / Virus Database: 2433/5076 - Release Date: 06/17/12
>

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From B.Tietz at pinguin.ag  Fri Aug  3 14:48:48 2012
From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag)
Date: Fri, 3 Aug 2012 12:48:48 +0200
Subject: [Freeswitch-users] Remove Privacy: none Header in INVITE
In-Reply-To: <20120803102413.GA28145@eagle.cupis.co.uk>
References: <07BF4904977CC645B485E970424193AD118F3F7BBF@localhost>
	<20120803102413.GA28145@eagle.cupis.co.uk>
Message-ID: <07BF4904977CC645B485E970424193AD118F3F7CC5@localhost>

Hi,

this is what I came up my mind right now too... :-)

Benjamin


-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Paul Cupis
Gesendet: Freitag, 3. August 2012 12:24
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Remove Privacy: none Header in INVITE

On Fri, Aug 03, 2012 at 11:09:24AM +0200, B.Tietz at pinguin.ag wrote:
> I have to send an INVITE with P-asserted-identity but without the Privacy: none header.
> Right now I use
> 
> <action application="bridge" 
> data="{sip_contact_user=12345,sip_cid_type=pid}sofia/gateway/gw1/..."/
> >
> 
> Is there anything I can set/unset to get this!?

This requirement from your carrier(?) seems a little odd, but you can do it "the hard way" with something like this:

  <action application="set" data="sip_cid_type=none"/>
  <action application="unset" data="sip_h_Privacy"/>
  <action application="unset" data="sip_h_P-Asserted-Identity"/>
  <action application="set" data="sip_h_P-Asserted-Identity=&quot;NAME&quot; <sip:NUMBER at IP_ADDRESS>"/>

I don't know an easier way to just say "send P-A-ID, but not Privacy".

Regards,


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From evgeniy at bestnet.kharkov.ua  Fri Aug  3 14:56:36 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Fri, 03 Aug 2012 13:56:36 +0300
Subject: [Freeswitch-users] Nibble Billing deducted balance from account
 even call doesn't come to my mobile number
In-Reply-To: <CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>	<50178DB7.7000606@bestnet.kharkov.ua>	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>	<50194496.10804@bestnet.kharkov.ua>	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>	<501A5AF4.9020008@bestnet.kharkov.ua>	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>	<501A6085.2020308@bestnet.kharkov.ua>
	<CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>
Message-ID: <501BAE64.3020304@bestnet.kharkov.ua>

I checked it twice from different computers and it works fine =\
I made another one paste - http://pastebin.com/iW9DaAWN
Can you show me how you make calls from lua script?

03.08.2012 11:41, SamyGo ?????:
> Hi
> Your link isn't working for me !!
>
> On Thu, Aug 2, 2012 at 4:12 PM, Evgeniy Movlyan
> <evgeniy at bestnet.kharkov.ua>wrote:
>
>> http://pastebin.com/enEmLVJz
>>
>> 02.08.2012 13:58, SamyGo ?????:
>>> Ok,
>>> Paste the whole call console log and share via pastebin.
>>> Maybe find some other thing breaking it.
>>> BR
>>> Sammy
>>> On Aug 2, 2012 3:52 PM, "Evgeniy Movlyan"<evgeniy at bestnet.kharkov.ua>
>>> wrote:
>>>
>>>> I changed my dialplan, but got the same message =(
>>>>
>>>> 02.08.2012 07:49, SamyGo ?????:
>>>>> You might wanna change your dialplan to something like this:
>>>>>
>>>>> <action application="bridge" data="{
>>>>> ignore_early_media=true
>>>> }sofia/gateway/${default_gateway}/${dialed_number}"/>
>>>>> That's How I;ve done it, no wonder I use LUA but the dial-string is
>>>> exactly
>>>>> as below.
>>>>>
>>>>> I think there is a difference if you do it like above, It will set the
>>>>> ignore_early_media on the B-leg session and  not on A-leg.
>>>>>
>>>>> This a snippet from the FS-1.6 book: *page:186*
>>>>>
>>>>> "Curly brackets are used "globally" for the duration of a call. Take
>> the
>>>>> following example where we are bridging a call to Darren's cell phone,
>>>>> 203-829-3150. We
>>>>> only want to ring the phone for 20 seconds, to avoid hitting voicemail.
>>>>>
>>>>> <application action="bridge"
>>>>> data="{call_timeout=20}sofia/my_provider/2038293150">
>>>>>
>>>>> *The variable in brackets is utilized on the newly setup channel,
>>>>> sofia/my_provider/2038293150.*"
>>>>>
>>>>>
>>>>> Regards,
>>>>> Sammy
>>>>>
>>>>>
>>>>> On Wed, Aug 1, 2012 at 8:00 PM, Evgeniy Movlyan
>>>>> <evgeniy at bestnet.kharkov.ua>wrote:
>>>>>
>>>>>> It did not work for me, i got the same message.
>>>>>> My outbond extension:
>>>>>>
>>>>>> <extension name="AUTH_rad">
>>>>>>      <condition field="${toll_allow}" expression="international"/>
>>>>>>       <condition field="${AUTH_RESULT_RADIUS}" expression="^NOK$"
>>>>>> break="on-true">
>>>>>>        <action application="lua" data="callfailed.lua
>>>> ${caller_id_number}"/>
>>>>>>       </condition>
>>>>>>       <condition field="${AUTH_RESULT_RADIUS}" expression="^OK$">
>>>>>>        <action application="set"
>>>>>> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>>>>>>        <action application="set"
>>>>>> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>>>>>>        <action application="set" data="ignore_early_media=true"/>
>>>>>>        <action application="bridge"
>>>>>> data="sofia/gateway/${default_gateway}/${dialed_number}"/>
>>>>>>       </condition>
>>>>>> </extension>
>>>>>>
>>>>>>
>>>>>> 31.07.2012 11:06, SamyGo ?????:
>>>>>>> Hi,
>>>>>>> Well its works perfect for me, do you guys have ignore_early_media
>> set
>>>> in
>>>>>>> your outbound string, if no then set it and then see what happens.
>>>>>>> On Jul 31, 2012 12:50 PM, "Evgeniy Movlyan"<
>> evgeniy at bestnet.kharkov.ua
>>>>>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> I have the same problem. When i am calling from one my extension to
>>>>>>>> another all is ok, but when i am calling to external number i got
>> this
>>>>>>>> message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is not
>> in
>>>>>>>> answered state".
>>>>>>>>
>>>>>>>> 31.07.2012 10:35, virendra bhati ?????:
>>>>>>>>> mod_nibblebill.c:465 Not billing
>>>>>>>>> 97183008 - call is not in answered state
>>>>>>>>
>>>>>>>> --
>>>>>>>> Evgeniy Movlyan,
>>>>>>>> BestNet Ltd.
>>>>>>>>
>>>>>>>>
>>>>>>
>>>>
>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>
>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>> --
>>>>>> Evgeniy Movlyan,
>>>>>> BestNet Ltd.
>>>>>>
>>>>>>
>>>>
>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>> --
>>>> Evgeniy Movlyan,
>>>> BestNet Ltd.
>>>>
>>>>
>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> --
>> Evgeniy Movlyan,
>> BestNet Ltd.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Evgeniy Movlyan,
BestNet Ltd.


From david.villasmil.work at gmail.com  Fri Aug  3 15:13:31 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 3 Aug 2012 13:13:31 +0200
Subject: [Freeswitch-users] ESL Log to console
In-Reply-To: <CAFGRPVreuL_rZRo4D=uNKXTm-RJ+We52vfFiPjbiThGZGH0M3g@mail.gmail.com>
References: <CAFGRPVpu_3t1JG8SPqEb8J9JBn6VYOXL1Aa79u_7Hj-uO0WB0A@mail.gmail.com>
	<CAKzWOxWT5J=HFJ8hB=bpo32ACtmmqrNYqNGSar=2incVFWej+w@mail.gmail.com>
	<CAFGRPVrV1UUunYtVs68+EX_PXn-yDGxEZhN3AJEBJY_KYoDcCg@mail.gmail.com>
	<CAKzWOxU+nY6Q-xfWd2bd8EUQYsefMpgSeR6-gDH=qx8+Vg4QiA@mail.gmail.com>
	<CAFGRPVqOfe-507gV1JwBGWc2BijFp50-Vr8_u2QzM8d8DrfS=Q@mail.gmail.com>
	<CAKzWOxWS98-B5q1rVO3eE0GR-i=PtO-biHDgDCULmOE9n5eRYw@mail.gmail.com>
	<CAFGRPVreuL_rZRo4D=uNKXTm-RJ+We52vfFiPjbiThGZGH0M3g@mail.gmail.com>
Message-ID: <CAFGRPVpu-566if4Zd+cXJ4hQQ1W4aqc8UVrpsxOC8Et1crZuNQ@mail.gmail.com>

Any thoughts?

On Thu, Aug 2, 2012 at 9:21 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Actually no. I'm writing a script to control FS, incoming calls will be
> screened and routing will de done via this script... I already have this in
> Lua, but everyting a call comes in a new script is lunched, and I don't
> want that anymore. So I'm trying to write an ESL script wich will connect
> to FS via ESL and receive events, act accordingly and instruct fs what to
> do, in terms of routing, channel variables definition (for later use in
> xml_curl_cdr, etc.
>
> Maybe my approach is not the correct one...
>
> Thanks for your help...
>
> David
>
> On Thu, Aug 2, 2012 at 9:07 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Well, it all depends on how you're calling this script. Is it being
>> called from the dialplan?
>> -MC
>>
>>
>> On Thu, Aug 2, 2012 at 11:25 AM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Ouch!
>>>
>>> Ok, this is not like in Lua... Ok, I'll try that.... is there any
>>> documentation regarding what can and can't be set and the conditions for
>>> setting them? What about bridging channels, etc?
>>>
>>> :(
>>>
>>> David
>>>
>>> On Thu, Aug 2, 2012 at 8:19 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Can't use execute and set unless this is an outbound ESL connection,
>>>> which it seems not to be. I'd try this:
>>>>
>>>> $con->api("uuid_setvar","$uuid customer_company $local_variable");
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Thu, Aug 2, 2012 at 11:01 AM, David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>>> Hello Michael,
>>>>>
>>>>> Yes, there's a call in progress. Executing it like you said worked.
>>>>> (api)
>>>>>
>>>>>
>>>>> Now, I'm trying to set a variable like so:
>>>>>
>>>>> $con->execute("set","customer_company             = " .
>>>>>  $local_variable , $uuid );
>>>>>
>>>>> But it doesn't get set...
>>>>>
>>>>> :(
>>>>>
>>>>> Thanks for your help!
>>>>>
>>>>> David
>>>>>
>>>>>
>>>>> On Thu, Aug 2, 2012 at 6:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> David,
>>>>>>
>>>>>> Do you have a call in progress at this point? If not then you'll need
>>>>>> to supply a uuid of a live call, as mentioned here:
>>>>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute
>>>>>>
>>>>>> Remember, "execute" means "execute dialplan application" so if
>>>>>> there's no channel then "execute" really doesn't mean a whole lot.
>>>>>>
>>>>>> Alternatively you could try something like this:
>>>>>>
>>>>>> $con->api("log","WARNING Don't cross the streams!!");
>>>>>>
>>>>>> Remember this rule of thumb: pretty much anything you type at fs_cli
>>>>>> is an "API" and therefore you can use $con->api(), whereas anything that is
>>>>>> a diaplan application requires an actual channel on which to run
>>>>>> $con->execute().
>>>>>>
>>>>>> Hope that makes sense... :)
>>>>>>
>>>>>> -Michael
>>>>>>
>>>>>> On Thu, Aug 2, 2012 at 8:43 AM, David Villasmil <
>>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>>
>>>>>>> Hello Guys,
>>>>>>>
>>>>>>> I'm starting off with ESL, which is cool, but I'm trying to log to
>>>>>>> the console like Lua's "freeswitch.consoleLog("info", "BLAH BLAH");" doing
>>>>>>> it like:
>>>>>>>
>>>>>>> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>>>>>>> $con->execute("log", "1, BlahBlah");
>>>>>>>
>>>>>>>
>>>>>>> But nothing gets in the log files or console... and I can't find any
>>>>>>> documentation as to how to log using "execute"...
>>>>>>>
>>>>>>> any ideas?
>>>>>>>
>>>>>>> Thanks!
>>>>>>>
>>>>>>> David
>>>>>>>
>>>>>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From govoiper at gmail.com  Fri Aug  3 15:15:33 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 3 Aug 2012 16:15:33 +0500
Subject: [Freeswitch-users] Nibble Billing deducted balance from account
 even call doesn't come to my mobile number
In-Reply-To: <501BAE64.3020304@bestnet.kharkov.ua>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<50194496.10804@bestnet.kharkov.ua>
	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>
	<501A5AF4.9020008@bestnet.kharkov.ua>
	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>
	<501A6085.2020308@bestnet.kharkov.ua>
	<CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>
	<501BAE64.3020304@bestnet.kharkov.ua>
Message-ID: <CAJUJwtgHFyrk7ybxAmH2xmaz=d=OgCy5uHOmGhnyOK3QpdRjRw@mail.gmail.com>

yes why not,
wait for some time while I access my files.

On Fri, Aug 3, 2012 at 3:56 PM, Evgeniy Movlyan
<evgeniy at bestnet.kharkov.ua>wrote:

> I checked it twice from different computers and it works fine =\
> I made another one paste - http://pastebin.com/iW9DaAWN
> Can you show me how you make calls from lua script?
>
> 03.08.2012 11:41, SamyGo ?????:
> > Hi
> > Your link isn't working for me !!
> >
> > On Thu, Aug 2, 2012 at 4:12 PM, Evgeniy Movlyan
> > <evgeniy at bestnet.kharkov.ua>wrote:
> >
> >> http://pastebin.com/enEmLVJz
> >>
> >> 02.08.2012 13:58, SamyGo ?????:
> >>> Ok,
> >>> Paste the whole call console log and share via pastebin.
> >>> Maybe find some other thing breaking it.
> >>> BR
> >>> Sammy
> >>> On Aug 2, 2012 3:52 PM, "Evgeniy Movlyan"<evgeniy at bestnet.kharkov.ua>
> >>> wrote:
> >>>
> >>>> I changed my dialplan, but got the same message =(
> >>>>
> >>>> 02.08.2012 07:49, SamyGo ?????:
> >>>>> You might wanna change your dialplan to something like this:
> >>>>>
> >>>>> <action application="bridge" data="{
> >>>>> ignore_early_media=true
> >>>> }sofia/gateway/${default_gateway}/${dialed_number}"/>
> >>>>> That's How I;ve done it, no wonder I use LUA but the dial-string is
> >>>> exactly
> >>>>> as below.
> >>>>>
> >>>>> I think there is a difference if you do it like above, It will set
> the
> >>>>> ignore_early_media on the B-leg session and  not on A-leg.
> >>>>>
> >>>>> This a snippet from the FS-1.6 book: *page:186*
> >>>>>
> >>>>> "Curly brackets are used "globally" for the duration of a call. Take
> >> the
> >>>>> following example where we are bridging a call to Darren's cell
> phone,
> >>>>> 203-829-3150. We
> >>>>> only want to ring the phone for 20 seconds, to avoid hitting
> voicemail.
> >>>>>
> >>>>> <application action="bridge"
> >>>>> data="{call_timeout=20}sofia/my_provider/2038293150">
> >>>>>
> >>>>> *The variable in brackets is utilized on the newly setup channel,
> >>>>> sofia/my_provider/2038293150.*"
> >>>>>
> >>>>>
> >>>>> Regards,
> >>>>> Sammy
> >>>>>
> >>>>>
> >>>>> On Wed, Aug 1, 2012 at 8:00 PM, Evgeniy Movlyan
> >>>>> <evgeniy at bestnet.kharkov.ua>wrote:
> >>>>>
> >>>>>> It did not work for me, i got the same message.
> >>>>>> My outbond extension:
> >>>>>>
> >>>>>> <extension name="AUTH_rad">
> >>>>>>      <condition field="${toll_allow}" expression="international"/>
> >>>>>>       <condition field="${AUTH_RESULT_RADIUS}" expression="^NOK$"
> >>>>>> break="on-true">
> >>>>>>        <action application="lua" data="callfailed.lua
> >>>> ${caller_id_number}"/>
> >>>>>>       </condition>
> >>>>>>       <condition field="${AUTH_RESULT_RADIUS}" expression="^OK$">
> >>>>>>        <action application="set"
> >>>>>> data="effective_caller_id_number=${outbound_caller_id_number}"/>
> >>>>>>        <action application="set"
> >>>>>> data="effective_caller_id_name=${outbound_caller_id_name}"/>
> >>>>>>        <action application="set" data="ignore_early_media=true"/>
> >>>>>>        <action application="bridge"
> >>>>>> data="sofia/gateway/${default_gateway}/${dialed_number}"/>
> >>>>>>       </condition>
> >>>>>> </extension>
> >>>>>>
> >>>>>>
> >>>>>> 31.07.2012 11:06, SamyGo ?????:
> >>>>>>> Hi,
> >>>>>>> Well its works perfect for me, do you guys have ignore_early_media
> >> set
> >>>> in
> >>>>>>> your outbound string, if no then set it and then see what happens.
> >>>>>>> On Jul 31, 2012 12:50 PM, "Evgeniy Movlyan"<
> >> evgeniy at bestnet.kharkov.ua
> >>>>>
> >>>>>>> wrote:
> >>>>>>>
> >>>>>>>> I have the same problem. When i am calling from one my extension
> to
> >>>>>>>> another all is ok, but when i am calling to external number i got
> >> this
> >>>>>>>> message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is
> not
> >> in
> >>>>>>>> answered state".
> >>>>>>>>
> >>>>>>>> 31.07.2012 10:35, virendra bhati ?????:
> >>>>>>>>> mod_nibblebill.c:465 Not billing
> >>>>>>>>> 97183008 - call is not in answered state
> >>>>>>>>
> >>>>>>>> --
> >>>>>>>> Evgeniy Movlyan,
> >>>>>>>> BestNet Ltd.
> >>>>>>>>
> >>>>>>>>
> >>>>>>
> >>>>
> >>
> _________________________________________________________________________
> >>>>>>>> Professional FreeSWITCH Consulting Services:
> >>>>>>>> consulting at freeswitch.org
> >>>>>>>> http://www.freeswitchsolutions.com
> >>>>>>>>
> >>>>>>>> 
> >>>>>>>> 
> >>>>>>>>
> >>>>>>>> Official FreeSWITCH Sites
> >>>>>>>> http://www.freeswitch.org
> >>>>>>>> http://wiki.freeswitch.org
> >>>>>>>> http://www.cluecon.com
> >>>>>>>>
> >>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
> >>>>>>>>
> >>>>>>>> FreeSWITCH-users mailing list
> >>>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>>> UNSUBSCRIBE:
> >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>>> http://www.freeswitch.org
> >>>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>
> >>
> _________________________________________________________________________
> >>>>>>> Professional FreeSWITCH Consulting Services:
> >>>>>>> consulting at freeswitch.org
> >>>>>>> http://www.freeswitchsolutions.com
> >>>>>>>
> >>>>>>> 
> >>>>>>> 
> >>>>>>>
> >>>>>>> Official FreeSWITCH Sites
> >>>>>>> http://www.freeswitch.org
> >>>>>>> http://wiki.freeswitch.org
> >>>>>>> http://www.cluecon.com
> >>>>>>>
> >>>>>>> Join Us At ClueCon - Aug 7-9, 2012
> >>>>>>>
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> UNSUBSCRIBE:
> >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> http://www.freeswitch.org
> >>>>>>
> >>>>>> --
> >>>>>> Evgeniy Movlyan,
> >>>>>> BestNet Ltd.
> >>>>>>
> >>>>>>
> >>>>
> >>
> _________________________________________________________________________
> >>>>>> Professional FreeSWITCH Consulting Services:
> >>>>>> consulting at freeswitch.org
> >>>>>> http://www.freeswitchsolutions.com
> >>>>>>
> >>>>>> 
> >>>>>> 
> >>>>>>
> >>>>>> Official FreeSWITCH Sites
> >>>>>> http://www.freeswitch.org
> >>>>>> http://wiki.freeswitch.org
> >>>>>> http://www.cluecon.com
> >>>>>>
> >>>>>> Join Us At ClueCon - Aug 7-9, 2012
> >>>>>>
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>> UNSUBSCRIBE:
> >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>> http://www.freeswitch.org
> >>>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>
> _________________________________________________________________________
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>>
> >>>>> 
> >>>>> 
> >>>>>
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://wiki.freeswitch.org
> >>>>> http://www.cluecon.com
> >>>>>
> >>>>> Join Us At ClueCon - Aug 7-9, 2012
> >>>>>
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>> --
> >>>> Evgeniy Movlyan,
> >>>> BestNet Ltd.
> >>>>
> >>>>
> >>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> Join Us At ClueCon - Aug 7-9, 2012
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> Join Us At ClueCon - Aug 7-9, 2012
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >> --
> >> Evgeniy Movlyan,
> >> BestNet Ltd.
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Evgeniy Movlyan,
> BestNet Ltd.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Aug  3 15:34:05 2012
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 03 Aug 2012 19:34:05 +0800
Subject: [Freeswitch-users] Fax (RX): Missing name/header variable in
 socket!?
In-Reply-To: <CALaOL+hYAxkVTEOLBCpCZB5H6LbZaLH_hW-UiagvP_XYAKOQbQ@mail.gmail.com>
References: <CALaOL+hYAxkVTEOLBCpCZB5H6LbZaLH_hW-UiagvP_XYAKOQbQ@mail.gmail.com>
Message-ID: <501BB72D.7070106@coppice.org>

On 08/02/2012 05:26 PM, Dennis wrote:
> hi,
>
> we are receiving many faxes (RX), but we are missing a variable in the socket.
>
> we do get the "remote station id", which is the fax-number itself, but
> we need the name/header of the incoming fax (for example: COMPANYNAME
> INC.).
>
There is no company name field in a FAX. You only get the remote station 
ID, and half the time even that is blank. You might find the name of the 
sender embedded at the top of each page in graphic form, and then again 
you might not.

Regards,
Steve



From qasimakhan at gmail.com  Fri Aug  3 15:36:55 2012
From: qasimakhan at gmail.com (qasimakhan at gmail.com)
Date: Fri, 3 Aug 2012 16:36:55 +0500
Subject: [Freeswitch-users] Nibble Billing deducted balance from account
 even call doesn't come to my mobile number
In-Reply-To: <CAJUJwtgHFyrk7ybxAmH2xmaz=d=OgCy5uHOmGhnyOK3QpdRjRw@mail.gmail.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<50194496.10804@bestnet.kharkov.ua>
	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>
	<501A5AF4.9020008@bestnet.kharkov.ua>
	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>
	<501A6085.2020308@bestnet.kharkov.ua>
	<CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>
	<501BAE64.3020304@bestnet.kharkov.ua>
	<CAJUJwtgHFyrk7ybxAmH2xmaz=d=OgCy5uHOmGhnyOK3QpdRjRw@mail.gmail.com>
Message-ID: <CAPBKugskuRA2FgE6H+fUsHrU4+N-E+2erantqShSKn7H6Qw99g@mail.gmail.com>

Try using pastebin.pk. It should work!


On Fri, Aug 3, 2012 at 4:15 PM, SamyGo <govoiper at gmail.com> wrote:

> yes why not,
> wait for some time while I access my files.
>
>
> On Fri, Aug 3, 2012 at 3:56 PM, Evgeniy Movlyan <
> evgeniy at bestnet.kharkov.ua> wrote:
>
>> I checked it twice from different computers and it works fine =\
>> I made another one paste - http://pastebin.com/iW9DaAWN
>> Can you show me how you make calls from lua script?
>>
>> 03.08.2012 11:41, SamyGo ?????:
>> > Hi
>> > Your link isn't working for me !!
>> >
>> > On Thu, Aug 2, 2012 at 4:12 PM, Evgeniy Movlyan
>> > <evgeniy at bestnet.kharkov.ua>wrote:
>> >
>> >> http://pastebin.com/enEmLVJz
>> >>
>> >> 02.08.2012 13:58, SamyGo ?????:
>> >>> Ok,
>> >>> Paste the whole call console log and share via pastebin.
>> >>> Maybe find some other thing breaking it.
>> >>> BR
>> >>> Sammy
>> >>> On Aug 2, 2012 3:52 PM, "Evgeniy Movlyan"<evgeniy at bestnet.kharkov.ua>
>> >>> wrote:
>> >>>
>> >>>> I changed my dialplan, but got the same message =(
>> >>>>
>> >>>> 02.08.2012 07:49, SamyGo ?????:
>> >>>>> You might wanna change your dialplan to something like this:
>> >>>>>
>> >>>>> <action application="bridge" data="{
>> >>>>> ignore_early_media=true
>> >>>> }sofia/gateway/${default_gateway}/${dialed_number}"/>
>> >>>>> That's How I;ve done it, no wonder I use LUA but the dial-string is
>> >>>> exactly
>> >>>>> as below.
>> >>>>>
>> >>>>> I think there is a difference if you do it like above, It will set
>> the
>> >>>>> ignore_early_media on the B-leg session and  not on A-leg.
>> >>>>>
>> >>>>> This a snippet from the FS-1.6 book: *page:186*
>> >>>>>
>> >>>>> "Curly brackets are used "globally" for the duration of a call. Take
>> >> the
>> >>>>> following example where we are bridging a call to Darren's cell
>> phone,
>> >>>>> 203-829-3150. We
>> >>>>> only want to ring the phone for 20 seconds, to avoid hitting
>> voicemail.
>> >>>>>
>> >>>>> <application action="bridge"
>> >>>>> data="{call_timeout=20}sofia/my_provider/2038293150">
>> >>>>>
>> >>>>> *The variable in brackets is utilized on the newly setup channel,
>> >>>>> sofia/my_provider/2038293150.*"
>> >>>>>
>> >>>>>
>> >>>>> Regards,
>> >>>>> Sammy
>> >>>>>
>> >>>>>
>> >>>>> On Wed, Aug 1, 2012 at 8:00 PM, Evgeniy Movlyan
>> >>>>> <evgeniy at bestnet.kharkov.ua>wrote:
>> >>>>>
>> >>>>>> It did not work for me, i got the same message.
>> >>>>>> My outbond extension:
>> >>>>>>
>> >>>>>> <extension name="AUTH_rad">
>> >>>>>>      <condition field="${toll_allow}" expression="international"/>
>> >>>>>>       <condition field="${AUTH_RESULT_RADIUS}" expression="^NOK$"
>> >>>>>> break="on-true">
>> >>>>>>        <action application="lua" data="callfailed.lua
>> >>>> ${caller_id_number}"/>
>> >>>>>>       </condition>
>> >>>>>>       <condition field="${AUTH_RESULT_RADIUS}" expression="^OK$">
>> >>>>>>        <action application="set"
>> >>>>>> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>> >>>>>>        <action application="set"
>> >>>>>> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>> >>>>>>        <action application="set" data="ignore_early_media=true"/>
>> >>>>>>        <action application="bridge"
>> >>>>>> data="sofia/gateway/${default_gateway}/${dialed_number}"/>
>> >>>>>>       </condition>
>> >>>>>> </extension>
>> >>>>>>
>> >>>>>>
>> >>>>>> 31.07.2012 11:06, SamyGo ?????:
>> >>>>>>> Hi,
>> >>>>>>> Well its works perfect for me, do you guys have ignore_early_media
>> >> set
>> >>>> in
>> >>>>>>> your outbound string, if no then set it and then see what happens.
>> >>>>>>> On Jul 31, 2012 12:50 PM, "Evgeniy Movlyan"<
>> >> evgeniy at bestnet.kharkov.ua
>> >>>>>
>> >>>>>>> wrote:
>> >>>>>>>
>> >>>>>>>> I have the same problem. When i am calling from one my extension
>> to
>> >>>>>>>> another all is ok, but when i am calling to external number i got
>> >> this
>> >>>>>>>> message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is
>> not
>> >> in
>> >>>>>>>> answered state".
>> >>>>>>>>
>> >>>>>>>> 31.07.2012 10:35, virendra bhati ?????:
>> >>>>>>>>> mod_nibblebill.c:465 Not billing
>> >>>>>>>>> 97183008 - call is not in answered state
>> >>>>>>>>
>> >>>>>>>> --
>> >>>>>>>> Evgeniy Movlyan,
>> >>>>>>>> BestNet Ltd.
>> >>>>>>>>
>> >>>>>>>>
>> >>>>>>
>> >>>>
>> >>
>> _________________________________________________________________________
>> >>>>>>>> Professional FreeSWITCH Consulting Services:
>> >>>>>>>> consulting at freeswitch.org
>> >>>>>>>> http://www.freeswitchsolutions.com
>> >>>>>>>>
>> >>>>>>>> 
>> >>>>>>>> 
>> >>>>>>>>
>> >>>>>>>> Official FreeSWITCH Sites
>> >>>>>>>> http://www.freeswitch.org
>> >>>>>>>> http://wiki.freeswitch.org
>> >>>>>>>> http://www.cluecon.com
>> >>>>>>>>
>> >>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>> >>>>>>>>
>> >>>>>>>> FreeSWITCH-users mailing list
>> >>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>>>> UNSUBSCRIBE:
>> >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>>>> http://www.freeswitch.org
>> >>>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>
>> >>
>> _________________________________________________________________________
>> >>>>>>> Professional FreeSWITCH Consulting Services:
>> >>>>>>> consulting at freeswitch.org
>> >>>>>>> http://www.freeswitchsolutions.com
>> >>>>>>>
>> >>>>>>> 
>> >>>>>>> 
>> >>>>>>>
>> >>>>>>> Official FreeSWITCH Sites
>> >>>>>>> http://www.freeswitch.org
>> >>>>>>> http://wiki.freeswitch.org
>> >>>>>>> http://www.cluecon.com
>> >>>>>>>
>> >>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>> >>>>>>>
>> >>>>>>> FreeSWITCH-users mailing list
>> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>>> UNSUBSCRIBE:
>> >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>>> http://www.freeswitch.org
>> >>>>>>
>> >>>>>> --
>> >>>>>> Evgeniy Movlyan,
>> >>>>>> BestNet Ltd.
>> >>>>>>
>> >>>>>>
>> >>>>
>> >>
>> _________________________________________________________________________
>> >>>>>> Professional FreeSWITCH Consulting Services:
>> >>>>>> consulting at freeswitch.org
>> >>>>>> http://www.freeswitchsolutions.com
>> >>>>>>
>> >>>>>> 
>> >>>>>> 
>> >>>>>>
>> >>>>>> Official FreeSWITCH Sites
>> >>>>>> http://www.freeswitch.org
>> >>>>>> http://wiki.freeswitch.org
>> >>>>>> http://www.cluecon.com
>> >>>>>>
>> >>>>>> Join Us At ClueCon - Aug 7-9, 2012
>> >>>>>>
>> >>>>>> FreeSWITCH-users mailing list
>> >>>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>> UNSUBSCRIBE:
>> >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>> http://www.freeswitch.org
>> >>>>>>
>> >>>>>
>> >>>>>
>> >>>>>
>> >>>>>
>> >>
>> _________________________________________________________________________
>> >>>>> Professional FreeSWITCH Consulting Services:
>> >>>>> consulting at freeswitch.org
>> >>>>> http://www.freeswitchsolutions.com
>> >>>>>
>> >>>>> 
>> >>>>> 
>> >>>>>
>> >>>>> Official FreeSWITCH Sites
>> >>>>> http://www.freeswitch.org
>> >>>>> http://wiki.freeswitch.org
>> >>>>> http://www.cluecon.com
>> >>>>>
>> >>>>> Join Us At ClueCon - Aug 7-9, 2012
>> >>>>>
>> >>>>> FreeSWITCH-users mailing list
>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>> UNSUBSCRIBE:
>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> http://www.freeswitch.org
>> >>>>
>> >>>> --
>> >>>> Evgeniy Movlyan,
>> >>>> BestNet Ltd.
>> >>>>
>> >>>>
>> >>
>> _________________________________________________________________________
>> >>>> Professional FreeSWITCH Consulting Services:
>> >>>> consulting at freeswitch.org
>> >>>> http://www.freeswitchsolutions.com
>> >>>>
>> >>>> 
>> >>>> 
>> >>>>
>> >>>> Official FreeSWITCH Sites
>> >>>> http://www.freeswitch.org
>> >>>> http://wiki.freeswitch.org
>> >>>> http://www.cluecon.com
>> >>>>
>> >>>> Join Us At ClueCon - Aug 7-9, 2012
>> >>>>
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>>
>> >>>
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://wiki.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> Join Us At ClueCon - Aug 7-9, 2012
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >> --
>> >> Evgeniy Movlyan,
>> >> BestNet Ltd.
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> Join Us At ClueCon - Aug 7-9, 2012
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > Join Us At ClueCon - Aug 7-9, 2012
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> --
>> Evgeniy Movlyan,
>> BestNet Ltd.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Aug  3 15:39:00 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 3 Aug 2012 13:39:00 +0200
Subject: [Freeswitch-users] ESL Log to console
In-Reply-To: <CAFGRPVpu-566if4Zd+cXJ4hQQ1W4aqc8UVrpsxOC8Et1crZuNQ@mail.gmail.com>
References: <CAFGRPVpu_3t1JG8SPqEb8J9JBn6VYOXL1Aa79u_7Hj-uO0WB0A@mail.gmail.com>
	<CAKzWOxWT5J=HFJ8hB=bpo32ACtmmqrNYqNGSar=2incVFWej+w@mail.gmail.com>
	<CAFGRPVrV1UUunYtVs68+EX_PXn-yDGxEZhN3AJEBJY_KYoDcCg@mail.gmail.com>
	<CAKzWOxU+nY6Q-xfWd2bd8EUQYsefMpgSeR6-gDH=qx8+Vg4QiA@mail.gmail.com>
	<CAFGRPVqOfe-507gV1JwBGWc2BijFp50-Vr8_u2QzM8d8DrfS=Q@mail.gmail.com>
	<CAKzWOxWS98-B5q1rVO3eE0GR-i=PtO-biHDgDCULmOE9n5eRYw@mail.gmail.com>
	<CAFGRPVreuL_rZRo4D=uNKXTm-RJ+We52vfFiPjbiThGZGH0M3g@mail.gmail.com>
	<CAFGRPVpu-566if4Zd+cXJ4hQQ1W4aqc8UVrpsxOC8Et1crZuNQ@mail.gmail.com>
Message-ID: <CAFGRPVqrhWYhxcf4O=KVbFbrTHs-44nDP6nYB5bOwsfCKHND7A@mail.gmail.com>

Hello,

Using ESL, at what point can one start sending commands to the channel?
It seems impossible for me to issue commands at

EVENT: CHANNEL_CREATE
STATE: CS_ROUTING

It maybe important to mention I don't have anything runing in the dialplan!
My script connects via ESL to FS and wait for events (incoming calls, etc)
and will issue commands like bridge the calls to their destination, etc.

Any help would be appreciated!

Thanks

David

On Fri, Aug 3, 2012 at 1:13 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Any thoughts?
>
>
> On Thu, Aug 2, 2012 at 9:21 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Actually no. I'm writing a script to control FS, incoming calls will be
>> screened and routing will de done via this script... I already have this in
>> Lua, but everyting a call comes in a new script is lunched, and I don't
>> want that anymore. So I'm trying to write an ESL script wich will connect
>> to FS via ESL and receive events, act accordingly and instruct fs what to
>> do, in terms of routing, channel variables definition (for later use in
>> xml_curl_cdr, etc.
>>
>> Maybe my approach is not the correct one...
>>
>> Thanks for your help...
>>
>> David
>>
>> On Thu, Aug 2, 2012 at 9:07 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Well, it all depends on how you're calling this script. Is it being
>>> called from the dialplan?
>>> -MC
>>>
>>>
>>> On Thu, Aug 2, 2012 at 11:25 AM, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> Ouch!
>>>>
>>>> Ok, this is not like in Lua... Ok, I'll try that.... is there any
>>>> documentation regarding what can and can't be set and the conditions for
>>>> setting them? What about bridging channels, etc?
>>>>
>>>> :(
>>>>
>>>> David
>>>>
>>>> On Thu, Aug 2, 2012 at 8:19 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> Can't use execute and set unless this is an outbound ESL connection,
>>>>> which it seems not to be. I'd try this:
>>>>>
>>>>> $con->api("uuid_setvar","$uuid customer_company $local_variable");
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>>> On Thu, Aug 2, 2012 at 11:01 AM, David Villasmil <
>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>
>>>>>> Hello Michael,
>>>>>>
>>>>>> Yes, there's a call in progress. Executing it like you said worked.
>>>>>> (api)
>>>>>>
>>>>>>
>>>>>> Now, I'm trying to set a variable like so:
>>>>>>
>>>>>> $con->execute("set","customer_company             = " .
>>>>>>  $local_variable , $uuid );
>>>>>>
>>>>>> But it doesn't get set...
>>>>>>
>>>>>> :(
>>>>>>
>>>>>> Thanks for your help!
>>>>>>
>>>>>> David
>>>>>>
>>>>>>
>>>>>> On Thu, Aug 2, 2012 at 6:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>
>>>>>>> David,
>>>>>>>
>>>>>>> Do you have a call in progress at this point? If not then you'll
>>>>>>> need to supply a uuid of a live call, as mentioned here:
>>>>>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute
>>>>>>>
>>>>>>> Remember, "execute" means "execute dialplan application" so if
>>>>>>> there's no channel then "execute" really doesn't mean a whole lot.
>>>>>>>
>>>>>>> Alternatively you could try something like this:
>>>>>>>
>>>>>>> $con->api("log","WARNING Don't cross the streams!!");
>>>>>>>
>>>>>>> Remember this rule of thumb: pretty much anything you type at fs_cli
>>>>>>> is an "API" and therefore you can use $con->api(), whereas anything that is
>>>>>>> a diaplan application requires an actual channel on which to run
>>>>>>> $con->execute().
>>>>>>>
>>>>>>> Hope that makes sense... :)
>>>>>>>
>>>>>>> -Michael
>>>>>>>
>>>>>>> On Thu, Aug 2, 2012 at 8:43 AM, David Villasmil <
>>>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>>>
>>>>>>>> Hello Guys,
>>>>>>>>
>>>>>>>> I'm starting off with ESL, which is cool, but I'm trying to log to
>>>>>>>> the console like Lua's "freeswitch.consoleLog("info", "BLAH BLAH");" doing
>>>>>>>> it like:
>>>>>>>>
>>>>>>>> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>>>>>>>> $con->execute("log", "1, BlahBlah");
>>>>>>>>
>>>>>>>>
>>>>>>>> But nothing gets in the log files or console... and I can't find
>>>>>>>> any documentation as to how to log using "execute"...
>>>>>>>>
>>>>>>>> any ideas?
>>>>>>>>
>>>>>>>> Thanks!
>>>>>>>>
>>>>>>>> David
>>>>>>>>
>>>>>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From odermann at googlemail.com  Fri Aug  3 17:03:39 2012
From: odermann at googlemail.com (Dennis)
Date: Fri, 3 Aug 2012 15:03:39 +0200
Subject: [Freeswitch-users] Fax (RX): Missing name/header variable in
	socket!?
In-Reply-To: <501BB72D.7070106@coppice.org>
References: <CALaOL+hYAxkVTEOLBCpCZB5H6LbZaLH_hW-UiagvP_XYAKOQbQ@mail.gmail.com>
	<501BB72D.7070106@coppice.org>
Message-ID: <CALaOL+h7-6pTtf0rVHD5X0wryawhyn7UZ5v87-mZ-jwXinE_gg@mail.gmail.com>

but with all fax-machines i know, one can enter a header, like a
company name, and this header will be seen on the incoming fax in the
head, most of the time at the side of the fax-number. if i receive a
fax with my fax-machine, i can often even see these header-information
in the display of my fax-machine, while receiving the fax.

where do those information come from and how can i access them?


From govoiper at gmail.com  Fri Aug  3 17:40:26 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 3 Aug 2012 18:40:26 +0500
Subject: [Freeswitch-users] Nibble Billing deducted balance from account
 even call doesn't come to my mobile number
In-Reply-To: <CAPBKugskuRA2FgE6H+fUsHrU4+N-E+2erantqShSKn7H6Qw99g@mail.gmail.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<50194496.10804@bestnet.kharkov.ua>
	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>
	<501A5AF4.9020008@bestnet.kharkov.ua>
	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>
	<501A6085.2020308@bestnet.kharkov.ua>
	<CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>
	<501BAE64.3020304@bestnet.kharkov.ua>
	<CAJUJwtgHFyrk7ybxAmH2xmaz=d=OgCy5uHOmGhnyOK3QpdRjRw@mail.gmail.com>
	<CAPBKugskuRA2FgE6H+fUsHrU4+N-E+2erantqShSKn7H6Qw99g@mail.gmail.com>
Message-ID: <CAJUJwti68vmSDFpAkwk08otnJyWHkOWJ6qRJ=AcStmtW=RT04Q@mail.gmail.com>

Here it is:
I had to delete alot of lines to make it understandable for you.


-- START OF CODE --
phone_number = argv[1];
client_id = argv[2];
B_PARTY = argv[3];
outbound_cli = argv[4];
rate = argv[5];

freeswitch.consoleLog("INFO","the number we are calling is " ..
phone_number.."\n");

session =
freeswitch.Session("{ignore_early_media=true,origination_caller_id_number="..outbound_cli.."}sofia/gateway/Asterisk/"
.. phone_number);
session:execute("nibblebill","heartbeat 30") -- triggered for 30 sec
billing deduction on A-Leg
session:setVariable("client_id",client_id)
session:setVariable("nibble_account",phone_number)
session:setVariable("nibble_rate",rate)
session:setVariable("nibble_increment",30)
if(session:answered() == true) then
freeswitch.consoleLog("INFO","Originate a New Call
for"..destination_data.." and Start Billing B-Leg too\n")
 dialB =
"{ignore_early_media=false,origination_caller_id_number="..B_PARTY.."}sofia/gateway/Asterisk/I"
legB = freeswitch.Session(dialB..destination_data,session)
legB:execute("nibblebill","heartbeat 60") -- Triggered for 60 sec billing
on B-Leg
legB:setVariable("client_id",client_id)
legB:setVariable("nibble_account",phone_number) --[[Deduct from A-Party's
cash]]--
legB:setVariable("nibble_rate",rate)
legB:setVariable("nibble_increment",30)
if ( legB:ready()) then
     freeswitch.consoleLog("NOTICE","------ALL READY BRIDGE THE
CALLS------\n")
freeswitch.bridge(legB,session)
else
   freeswitch.consoleLog("NOTICE","It appears that " ..session.. " or " ..
dialB .. " disconnected...\n")
  end
session:hangup();
 end
end
freeswitch.consoleLog("INFO","hangup reasons '"..session:hangupCause().."'
\n");
session:hangup();
---END OF CODE--



BR
Sammy


On Fri, Aug 3, 2012 at 4:36 PM, qasimakhan at gmail.com
<qasimakhan at gmail.com>wrote:

> Try using pastebin.pk. It should work!
>
>
>
> On Fri, Aug 3, 2012 at 4:15 PM, SamyGo <govoiper at gmail.com> wrote:
>
>> yes why not,
>> wait for some time while I access my files.
>>
>>
>> On Fri, Aug 3, 2012 at 3:56 PM, Evgeniy Movlyan <
>> evgeniy at bestnet.kharkov.ua> wrote:
>>
>>> I checked it twice from different computers and it works fine =\
>>> I made another one paste - http://pastebin.com/iW9DaAWN
>>> Can you show me how you make calls from lua script?
>>>
>>> 03.08.2012 11:41, SamyGo ?????:
>>> > Hi
>>> > Your link isn't working for me !!
>>> >
>>> > On Thu, Aug 2, 2012 at 4:12 PM, Evgeniy Movlyan
>>> > <evgeniy at bestnet.kharkov.ua>wrote:
>>> >
>>> >> http://pastebin.com/enEmLVJz
>>> >>
>>> >> 02.08.2012 13:58, SamyGo ?????:
>>> >>> Ok,
>>> >>> Paste the whole call console log and share via pastebin.
>>> >>> Maybe find some other thing breaking it.
>>> >>> BR
>>> >>> Sammy
>>> >>> On Aug 2, 2012 3:52 PM, "Evgeniy Movlyan"<evgeniy at bestnet.kharkov.ua
>>> >
>>> >>> wrote:
>>> >>>
>>> >>>> I changed my dialplan, but got the same message =(
>>> >>>>
>>> >>>> 02.08.2012 07:49, SamyGo ?????:
>>> >>>>> You might wanna change your dialplan to something like this:
>>> >>>>>
>>> >>>>> <action application="bridge" data="{
>>> >>>>> ignore_early_media=true
>>> >>>> }sofia/gateway/${default_gateway}/${dialed_number}"/>
>>> >>>>> That's How I;ve done it, no wonder I use LUA but the dial-string is
>>> >>>> exactly
>>> >>>>> as below.
>>> >>>>>
>>> >>>>> I think there is a difference if you do it like above, It will set
>>> the
>>> >>>>> ignore_early_media on the B-leg session and  not on A-leg.
>>> >>>>>
>>> >>>>> This a snippet from the FS-1.6 book: *page:186*
>>> >>>>>
>>> >>>>> "Curly brackets are used "globally" for the duration of a call.
>>> Take
>>> >> the
>>> >>>>> following example where we are bridging a call to Darren's cell
>>> phone,
>>> >>>>> 203-829-3150. We
>>> >>>>> only want to ring the phone for 20 seconds, to avoid hitting
>>> voicemail.
>>> >>>>>
>>> >>>>> <application action="bridge"
>>> >>>>> data="{call_timeout=20}sofia/my_provider/2038293150">
>>> >>>>>
>>> >>>>> *The variable in brackets is utilized on the newly setup channel,
>>> >>>>> sofia/my_provider/2038293150.*"
>>> >>>>>
>>> >>>>>
>>> >>>>> Regards,
>>> >>>>> Sammy
>>> >>>>>
>>> >>>>>
>>> >>>>> On Wed, Aug 1, 2012 at 8:00 PM, Evgeniy Movlyan
>>> >>>>> <evgeniy at bestnet.kharkov.ua>wrote:
>>> >>>>>
>>> >>>>>> It did not work for me, i got the same message.
>>> >>>>>> My outbond extension:
>>> >>>>>>
>>> >>>>>> <extension name="AUTH_rad">
>>> >>>>>>      <condition field="${toll_allow}" expression="international"/>
>>> >>>>>>       <condition field="${AUTH_RESULT_RADIUS}" expression="^NOK$"
>>> >>>>>> break="on-true">
>>> >>>>>>        <action application="lua" data="callfailed.lua
>>> >>>> ${caller_id_number}"/>
>>> >>>>>>       </condition>
>>> >>>>>>       <condition field="${AUTH_RESULT_RADIUS}" expression="^OK$">
>>> >>>>>>        <action application="set"
>>> >>>>>> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>>> >>>>>>        <action application="set"
>>> >>>>>> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>>> >>>>>>        <action application="set" data="ignore_early_media=true"/>
>>> >>>>>>        <action application="bridge"
>>> >>>>>> data="sofia/gateway/${default_gateway}/${dialed_number}"/>
>>> >>>>>>       </condition>
>>> >>>>>> </extension>
>>> >>>>>>
>>> >>>>>>
>>> >>>>>> 31.07.2012 11:06, SamyGo ?????:
>>> >>>>>>> Hi,
>>> >>>>>>> Well its works perfect for me, do you guys have
>>> ignore_early_media
>>> >> set
>>> >>>> in
>>> >>>>>>> your outbound string, if no then set it and then see what
>>> happens.
>>> >>>>>>> On Jul 31, 2012 12:50 PM, "Evgeniy Movlyan"<
>>> >> evgeniy at bestnet.kharkov.ua
>>> >>>>>
>>> >>>>>>> wrote:
>>> >>>>>>>
>>> >>>>>>>> I have the same problem. When i am calling from one my
>>> extension to
>>> >>>>>>>> another all is ok, but when i am calling to external number i
>>> got
>>> >> this
>>> >>>>>>>> message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is
>>> not
>>> >> in
>>> >>>>>>>> answered state".
>>> >>>>>>>>
>>> >>>>>>>> 31.07.2012 10:35, virendra bhati ?????:
>>> >>>>>>>>> mod_nibblebill.c:465 Not billing
>>> >>>>>>>>> 97183008 - call is not in answered state
>>> >>>>>>>>
>>> >>>>>>>> --
>>> >>>>>>>> Evgeniy Movlyan,
>>> >>>>>>>> BestNet Ltd.
>>> >>>>>>>>
>>> >>>>>>>>
>>> >>>>>>
>>> >>>>
>>> >>
>>> _________________________________________________________________________
>>> >>>>>>>> Professional FreeSWITCH Consulting Services:
>>> >>>>>>>> consulting at freeswitch.org
>>> >>>>>>>> http://www.freeswitchsolutions.com
>>> >>>>>>>>
>>> >>>>>>>> 
>>> >>>>>>>> 
>>> >>>>>>>>
>>> >>>>>>>> Official FreeSWITCH Sites
>>> >>>>>>>> http://www.freeswitch.org
>>> >>>>>>>> http://wiki.freeswitch.org
>>> >>>>>>>> http://www.cluecon.com
>>> >>>>>>>>
>>> >>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>> >>>>>>>>
>>> >>>>>>>> FreeSWITCH-users mailing list
>>> >>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>>>>> UNSUBSCRIBE:
>>> >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>>>>>> http://www.freeswitch.org
>>> >>>>>>>>
>>> >>>>>>>
>>> >>>>>>>
>>> >>>>>>>
>>> >>>>>>>
>>> >>>>
>>> >>
>>> _________________________________________________________________________
>>> >>>>>>> Professional FreeSWITCH Consulting Services:
>>> >>>>>>> consulting at freeswitch.org
>>> >>>>>>> http://www.freeswitchsolutions.com
>>> >>>>>>>
>>> >>>>>>> 
>>> >>>>>>> 
>>> >>>>>>>
>>> >>>>>>> Official FreeSWITCH Sites
>>> >>>>>>> http://www.freeswitch.org
>>> >>>>>>> http://wiki.freeswitch.org
>>> >>>>>>> http://www.cluecon.com
>>> >>>>>>>
>>> >>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>> >>>>>>>
>>> >>>>>>> FreeSWITCH-users mailing list
>>> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>>>> UNSUBSCRIBE:
>>> >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>>>>> http://www.freeswitch.org
>>> >>>>>>
>>> >>>>>> --
>>> >>>>>> Evgeniy Movlyan,
>>> >>>>>> BestNet Ltd.
>>> >>>>>>
>>> >>>>>>
>>> >>>>
>>> >>
>>> _________________________________________________________________________
>>> >>>>>> Professional FreeSWITCH Consulting Services:
>>> >>>>>> consulting at freeswitch.org
>>> >>>>>> http://www.freeswitchsolutions.com
>>> >>>>>>
>>> >>>>>> 
>>> >>>>>> 
>>> >>>>>>
>>> >>>>>> Official FreeSWITCH Sites
>>> >>>>>> http://www.freeswitch.org
>>> >>>>>> http://wiki.freeswitch.org
>>> >>>>>> http://www.cluecon.com
>>> >>>>>>
>>> >>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>> >>>>>>
>>> >>>>>> FreeSWITCH-users mailing list
>>> >>>>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>>> UNSUBSCRIBE:
>>> >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>>>> http://www.freeswitch.org
>>> >>>>>>
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> >>>>>
>>> >>
>>> _________________________________________________________________________
>>> >>>>> Professional FreeSWITCH Consulting Services:
>>> >>>>> consulting at freeswitch.org
>>> >>>>> http://www.freeswitchsolutions.com
>>> >>>>>
>>> >>>>> 
>>> >>>>> 
>>> >>>>>
>>> >>>>> Official FreeSWITCH Sites
>>> >>>>> http://www.freeswitch.org
>>> >>>>> http://wiki.freeswitch.org
>>> >>>>> http://www.cluecon.com
>>> >>>>>
>>> >>>>> Join Us At ClueCon - Aug 7-9, 2012
>>> >>>>>
>>> >>>>> FreeSWITCH-users mailing list
>>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>> UNSUBSCRIBE:
>>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>>> http://www.freeswitch.org
>>> >>>>
>>> >>>> --
>>> >>>> Evgeniy Movlyan,
>>> >>>> BestNet Ltd.
>>> >>>>
>>> >>>>
>>> >>
>>> _________________________________________________________________________
>>> >>>> Professional FreeSWITCH Consulting Services:
>>> >>>> consulting at freeswitch.org
>>> >>>> http://www.freeswitchsolutions.com
>>> >>>>
>>> >>>> 
>>> >>>> 
>>> >>>>
>>> >>>> Official FreeSWITCH Sites
>>> >>>> http://www.freeswitch.org
>>> >>>> http://wiki.freeswitch.org
>>> >>>> http://www.cluecon.com
>>> >>>>
>>> >>>> Join Us At ClueCon - Aug 7-9, 2012
>>> >>>>
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>> UNSUBSCRIBE:
>>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>>
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> _________________________________________________________________________
>>> >>> Professional FreeSWITCH Consulting Services:
>>> >>> consulting at freeswitch.org
>>> >>> http://www.freeswitchsolutions.com
>>> >>>
>>> >>> 
>>> >>> 
>>> >>>
>>> >>> Official FreeSWITCH Sites
>>> >>> http://www.freeswitch.org
>>> >>> http://wiki.freeswitch.org
>>> >>> http://www.cluecon.com
>>> >>>
>>> >>> Join Us At ClueCon - Aug 7-9, 2012
>>> >>>
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >> --
>>> >> Evgeniy Movlyan,
>>> >> BestNet Ltd.
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> 
>>> >> 
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://wiki.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> Join Us At ClueCon - Aug 7-9, 2012
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > Join Us At ClueCon - Aug 7-9, 2012
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> --
>>> Evgeniy Movlyan,
>>> BestNet Ltd.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From evgeniy at bestnet.kharkov.ua  Fri Aug  3 17:55:01 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Fri, 03 Aug 2012 16:55:01 +0300
Subject: [Freeswitch-users] Nibble Billing deducted balance from account
 even call doesn't come to my mobile number
In-Reply-To: <CAJUJwti68vmSDFpAkwk08otnJyWHkOWJ6qRJ=AcStmtW=RT04Q@mail.gmail.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>	<50178DB7.7000606@bestnet.kharkov.ua>	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>	<50194496.10804@bestnet.kharkov.ua>	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>	<501A5AF4.9020008@bestnet.kharkov.ua>	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>	<501A6085.2020308@bestnet.kharkov.ua>	<CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>	<501BAE64.3020304@bestnet.kharkov.ua>	<CAJUJwtgHFyrk7ybxAmH2xmaz=d=OgCy5uHOmGhnyOK3QpdRjRw@mail.gmail.com>	<CAPBKugskuRA2FgE6H+fUsHrU4+N-E+2erantqShSKn7H6Qw99g@mail.gmail.com>
	<CAJUJwti68vmSDFpAkwk08otnJyWHkOWJ6qRJ=AcStmtW=RT04Q@mail.gmail.com>
Message-ID: <501BD835.3060308@bestnet.kharkov.ua>

Thanks a lot.

03.08.2012 16:40, SamyGo ?????:
> Here it is:
> I had to delete alot of lines to make it understandable for you.
>
>
> -- START OF CODE --
> phone_number = argv[1];
> client_id = argv[2];
> B_PARTY = argv[3];
> outbound_cli = argv[4];
> rate = argv[5];
>
> freeswitch.consoleLog("INFO","the number we are calling is " ..
> phone_number.."\n");
>
> session =
> freeswitch.Session("{ignore_early_media=true,origination_caller_id_number="..outbound_cli.."}sofia/gateway/Asterisk/"
> .. phone_number);
> session:execute("nibblebill","heartbeat 30") -- triggered for 30 sec
> billing deduction on A-Leg
> session:setVariable("client_id",client_id)
> session:setVariable("nibble_account",phone_number)
> session:setVariable("nibble_rate",rate)
> session:setVariable("nibble_increment",30)
> if(session:answered() == true) then
> freeswitch.consoleLog("INFO","Originate a New Call
> for"..destination_data.." and Start Billing B-Leg too\n")
>   dialB =
> "{ignore_early_media=false,origination_caller_id_number="..B_PARTY.."}sofia/gateway/Asterisk/I"
> legB = freeswitch.Session(dialB..destination_data,session)
> legB:execute("nibblebill","heartbeat 60") -- Triggered for 60 sec billing
> on B-Leg
> legB:setVariable("client_id",client_id)
> legB:setVariable("nibble_account",phone_number) --[[Deduct from A-Party's
> cash]]--
> legB:setVariable("nibble_rate",rate)
> legB:setVariable("nibble_increment",30)
> if ( legB:ready()) then
>       freeswitch.consoleLog("NOTICE","------ALL READY BRIDGE THE
> CALLS------\n")
> freeswitch.bridge(legB,session)
> else
>     freeswitch.consoleLog("NOTICE","It appears that " ..session.. " or " ..
> dialB .. " disconnected...\n")
>    end
> session:hangup();
>   end
> end
> freeswitch.consoleLog("INFO","hangup reasons '"..session:hangupCause().."'
> \n");
> session:hangup();
> ---END OF CODE--
>
>
>
> BR
> Sammy
>
>
> On Fri, Aug 3, 2012 at 4:36 PM, qasimakhan at gmail.com
> <qasimakhan at gmail.com>wrote:
>
>> Try using pastebin.pk. It should work!
>>
>>
>>
>> On Fri, Aug 3, 2012 at 4:15 PM, SamyGo<govoiper at gmail.com>  wrote:
>>
>>> yes why not,
>>> wait for some time while I access my files.
>>>
>>>
>>> On Fri, Aug 3, 2012 at 3:56 PM, Evgeniy Movlyan<
>>> evgeniy at bestnet.kharkov.ua>  wrote:
>>>
>>>> I checked it twice from different computers and it works fine =\
>>>> I made another one paste - http://pastebin.com/iW9DaAWN
>>>> Can you show me how you make calls from lua script?
>>>>
>>>> 03.08.2012 11:41, SamyGo ?????:
>>>>> Hi
>>>>> Your link isn't working for me !!
>>>>>
>>>>> On Thu, Aug 2, 2012 at 4:12 PM, Evgeniy Movlyan
>>>>> <evgeniy at bestnet.kharkov.ua>wrote:
>>>>>
>>>>>> http://pastebin.com/enEmLVJz
>>>>>>
>>>>>> 02.08.2012 13:58, SamyGo ?????:
>>>>>>> Ok,
>>>>>>> Paste the whole call console log and share via pastebin.
>>>>>>> Maybe find some other thing breaking it.
>>>>>>> BR
>>>>>>> Sammy
>>>>>>> On Aug 2, 2012 3:52 PM, "Evgeniy Movlyan"<evgeniy at bestnet.kharkov.ua
>>>>>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> I changed my dialplan, but got the same message =(
>>>>>>>>
>>>>>>>> 02.08.2012 07:49, SamyGo ?????:
>>>>>>>>> You might wanna change your dialplan to something like this:
>>>>>>>>>
>>>>>>>>> <action application="bridge" data="{
>>>>>>>>> ignore_early_media=true
>>>>>>>> }sofia/gateway/${default_gateway}/${dialed_number}"/>
>>>>>>>>> That's How I;ve done it, no wonder I use LUA but the dial-string is
>>>>>>>> exactly
>>>>>>>>> as below.
>>>>>>>>>
>>>>>>>>> I think there is a difference if you do it like above, It will set
>>>> the
>>>>>>>>> ignore_early_media on the B-leg session and  not on A-leg.
>>>>>>>>>
>>>>>>>>> This a snippet from the FS-1.6 book: *page:186*
>>>>>>>>>
>>>>>>>>> "Curly brackets are used "globally" for the duration of a call.
>>>> Take
>>>>>> the
>>>>>>>>> following example where we are bridging a call to Darren's cell
>>>> phone,
>>>>>>>>> 203-829-3150. We
>>>>>>>>> only want to ring the phone for 20 seconds, to avoid hitting
>>>> voicemail.
>>>>>>>>>
>>>>>>>>> <application action="bridge"
>>>>>>>>> data="{call_timeout=20}sofia/my_provider/2038293150">
>>>>>>>>>
>>>>>>>>> *The variable in brackets is utilized on the newly setup channel,
>>>>>>>>> sofia/my_provider/2038293150.*"
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Regards,
>>>>>>>>> Sammy
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Wed, Aug 1, 2012 at 8:00 PM, Evgeniy Movlyan
>>>>>>>>> <evgeniy at bestnet.kharkov.ua>wrote:
>>>>>>>>>
>>>>>>>>>> It did not work for me, i got the same message.
>>>>>>>>>> My outbond extension:
>>>>>>>>>>
>>>>>>>>>> <extension name="AUTH_rad">
>>>>>>>>>>       <condition field="${toll_allow}" expression="international"/>
>>>>>>>>>>        <condition field="${AUTH_RESULT_RADIUS}" expression="^NOK$"
>>>>>>>>>> break="on-true">
>>>>>>>>>>         <action application="lua" data="callfailed.lua
>>>>>>>> ${caller_id_number}"/>
>>>>>>>>>>        </condition>
>>>>>>>>>>        <condition field="${AUTH_RESULT_RADIUS}" expression="^OK$">
>>>>>>>>>>         <action application="set"
>>>>>>>>>> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>>>>>>>>>>         <action application="set"
>>>>>>>>>> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>>>>>>>>>>         <action application="set" data="ignore_early_media=true"/>
>>>>>>>>>>         <action application="bridge"
>>>>>>>>>> data="sofia/gateway/${default_gateway}/${dialed_number}"/>
>>>>>>>>>>        </condition>
>>>>>>>>>> </extension>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> 31.07.2012 11:06, SamyGo ?????:
>>>>>>>>>>> Hi,
>>>>>>>>>>> Well its works perfect for me, do you guys have
>>>> ignore_early_media
>>>>>> set
>>>>>>>> in
>>>>>>>>>>> your outbound string, if no then set it and then see what
>>>> happens.
>>>>>>>>>>> On Jul 31, 2012 12:50 PM, "Evgeniy Movlyan"<
>>>>>> evgeniy at bestnet.kharkov.ua
>>>>>>>>>
>>>>>>>>>>> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> I have the same problem. When i am calling from one my
>>>> extension to
>>>>>>>>>>>> another all is ok, but when i am calling to external number i
>>>> got
>>>>>> this
>>>>>>>>>>>> message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is
>>>> not
>>>>>> in
>>>>>>>>>>>> answered state".
>>>>>>>>>>>>
>>>>>>>>>>>> 31.07.2012 10:35, virendra bhati ?????:
>>>>>>>>>>>>> mod_nibblebill.c:465 Not billing
>>>>>>>>>>>>> 97183008 - call is not in answered state
>>>>>>>>>>>>
>>>>>>>>>>>> --
>>>>>>>>>>>> Evgeniy Movlyan,
>>>>>>>>>>>> BestNet Ltd.
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>
>>>>>>
>>>> _________________________________________________________________________
>>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>>
>>>>>>>>>>>> 
>>>>>>>>>>>> 
>>>>>>>>>>>>
>>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>>> http://www.cluecon.com
>>>>>>>>>>>>
>>>>>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>>>>>
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>
>>>>>>
>>>> _________________________________________________________________________
>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>>
>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>> http://www.cluecon.com
>>>>>>>>>>>
>>>>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> Evgeniy Movlyan,
>>>>>>>>>> BestNet Ltd.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>
>>>>>>
>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.com
>>>>>>>>>>
>>>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>
>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>> --
>>>>>>>> Evgeniy Movlyan,
>>>>>>>> BestNet Ltd.
>>>>>>>>
>>>>>>>>
>>>>>>
>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>> --
>>>>>> Evgeniy Movlyan,
>>>>>> BestNet Ltd.
>>>>>>
>>>>>>
>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>> --
>>>> Evgeniy Movlyan,
>>>> BestNet Ltd.
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Evgeniy Movlyan,
BestNet Ltd.


From steveu at coppice.org  Fri Aug  3 18:36:22 2012
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 03 Aug 2012 22:36:22 +0800
Subject: [Freeswitch-users] Fax (RX): Missing name/header variable in
 socket!?
In-Reply-To: <CALaOL+h7-6pTtf0rVHD5X0wryawhyn7UZ5v87-mZ-jwXinE_gg@mail.gmail.com>
References: <CALaOL+hYAxkVTEOLBCpCZB5H6LbZaLH_hW-UiagvP_XYAKOQbQ@mail.gmail.com>
	<501BB72D.7070106@coppice.org>
	<CALaOL+h7-6pTtf0rVHD5X0wryawhyn7UZ5v87-mZ-jwXinE_gg@mail.gmail.com>
Message-ID: <501BE1E6.1030101@coppice.org>

On 08/03/2012 09:03 PM, Dennis wrote:
> but with all fax-machines i know, one can enter a header, like a
> company name, and this header will be seen on the incoming fax in the
> head, most of the time at the side of the fax-number. if i receive a
> fax with my fax-machine, i can often even see these header-information
> in the display of my fax-machine, while receiving the fax.
>
> where do those information come from and how can i access them?
>
See http://www.soft-switch.org/spandsp_faq/ar01s11.html and 
http://www.soft-switch.org/spandsp_faq/ar01s10.html

Steve



From gustavomarsico at gmail.com  Fri Aug  3 18:39:27 2012
From: gustavomarsico at gmail.com (ServTelar)
Date: Fri, 3 Aug 2012 11:39:27 -0300
Subject: [Freeswitch-users] Directory xml_curl, gets Can't find user
Message-ID: <3B0CFE1B-2F8F-428B-85DB-B420A2C5FE18@gmail.com>

Hi all

It seems this is a very old one, but certainly I didn't find any solution to this.

Using XML for directory everything works fine. But if I try using xml_curl, sending response:

<document type="freeswitch/xml">
  <section description="FreeSWITCH Directory" name="directory">
    <domain name="10.8.0.70">
      <params>
        <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
      </params>
      <groups>
        <group name="default">
          <users>
            <user id="2692128">
              <params>
                <param name="a1-hash" value="7d73f412ca332fedd81a297b64c0b271"/>
                <param name="vm-password" value="2128"/>
              </params>
              <variables>
                <variable name="accountcode" value="2692128"/>
                <variable name="user_context" value="default"/>
                <variable name="effective_caller_id_name" value="2692128"/>
                <variable name="effective_caller_id_number" value="2692128"/>
                <variable name="outbound_caller_id_name" value="2692128"/>
                <variable name="outbound_caller_id_number" value="2692128"/>
              </variables>
            </user>
          </users>
        </group>
      </groups>
    </domain>
  </section>
</document>

Registration is accepted:

Registrations:
=================================================================================================
Call-ID:    	939540755
User:       	2692128 at 10.8.0.70
Contact:    	2692128 <sip:2692128 at 10.0.0.10:5060;line=3dfb73927e26966;fs_nat=yes;fs_path=sip%3A2692128%4010.94.56.6%3A5060%3Bline%3D3dfb73927e26966>
Agent:      	iSoftPhone 3.4110
Status:     	Registered(UDP-NAT)(unknown) EXP(2012-08-03 11:35:30) EXPSECS(1688)
Host:       	demo
IP:         	10.94.56.6
Port:       	5060
Auth-User:  	2692128
Auth-Realm: 	10.8.0.70
MWI-Account:	2692128 at 10.8.0.70


But if I made a call pointing to that user I just got:

2012-08-03 11:05:38.261385 [WARNING] mod_dptools.c:3294 Can't find user [2692128 at 10.8.0.70]
2012-08-03 11:05:38.281377 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2012-08-03 11:05:38.281377 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT]

Domain are set both to the IP:

  <X-PRE-PROCESS cmd="set" data="domain=10.8.0.70"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=10.8.0.70"/>

And matches with the internal

freeswitch at internal> sofia status 
                     Name	   Type	                                      Data	State
=================================================================================================
                10.8.0.70	  alias	                                  internal	ALIASED
                 internal	profile	              sip:mod_sofia at 10.8.0.70:5060	RUNNING (0)


Can someone drop me a line to point where I should look to?


Thanks in advance

Gustavo




From mi.ke at null.net  Fri Aug  3 18:30:38 2012
From: mi.ke at null.net (Mi Ke)
Date: Fri, 03 Aug 2012 10:30:38 -0400
Subject: [Freeswitch-users] Early media: Sending 180 instead of 183?
Message-ID: <20120803143038.154610@gmx.com>

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



----- Original Message -----
From: Dennis
Sent: 08/03/12 12:32 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Early media: Sending 180 instead of 183?

 hi, we are doing a lot with early media for playing soundfiles/messages to the caller without charging the caller, while they have to wait. we are doing a "pre answer" -> fs sends a 183 message -> we start a playback -> fs lets flow the media. the problem is, that it happens sometimes, that the caller can not hear anything. the problem might be, that in germany the specification ITU-T Rec. Q.1912.5 does not necessarily map the 183 message to ISUP. in germany it seems to be the best way, to start a playback with a 180 message instead of a 183 message. is it possible to make fs sending a 180 message without making a bridge (because we do not always want to connect to another target)? thanks and kind regards dennis _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.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 Aug  3 19:26:58 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Aug 2012 08:26:58 -0700
Subject: [Freeswitch-users] ESL Log to console
In-Reply-To: <CAFGRPVqrhWYhxcf4O=KVbFbrTHs-44nDP6nYB5bOwsfCKHND7A@mail.gmail.com>
References: <CAFGRPVpu_3t1JG8SPqEb8J9JBn6VYOXL1Aa79u_7Hj-uO0WB0A@mail.gmail.com>
	<CAKzWOxWT5J=HFJ8hB=bpo32ACtmmqrNYqNGSar=2incVFWej+w@mail.gmail.com>
	<CAFGRPVrV1UUunYtVs68+EX_PXn-yDGxEZhN3AJEBJY_KYoDcCg@mail.gmail.com>
	<CAKzWOxU+nY6Q-xfWd2bd8EUQYsefMpgSeR6-gDH=qx8+Vg4QiA@mail.gmail.com>
	<CAFGRPVqOfe-507gV1JwBGWc2BijFp50-Vr8_u2QzM8d8DrfS=Q@mail.gmail.com>
	<CAKzWOxWS98-B5q1rVO3eE0GR-i=PtO-biHDgDCULmOE9n5eRYw@mail.gmail.com>
	<CAFGRPVreuL_rZRo4D=uNKXTm-RJ+We52vfFiPjbiThGZGH0M3g@mail.gmail.com>
	<CAFGRPVpu-566if4Zd+cXJ4hQQ1W4aqc8UVrpsxOC8Et1crZuNQ@mail.gmail.com>
	<CAFGRPVqrhWYhxcf4O=KVbFbrTHs-44nDP6nYB5bOwsfCKHND7A@mail.gmail.com>
Message-ID: <CAKzWOxX=dBU-5__+5kV_Tonyxrmas1pgT=Nr=B13SRMoOnXaSg@mail.gmail.com>

I think you need to pastebin your dialplan and script so we can see this in
context...
-MC

On Fri, Aug 3, 2012 at 4:39 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello,
>
> Using ESL, at what point can one start sending commands to the channel?
> It seems impossible for me to issue commands at
>
> EVENT: CHANNEL_CREATE
> STATE: CS_ROUTING
>
> It maybe important to mention I don't have anything runing in the dialplan!
> My script connects via ESL to FS and wait for events (incoming calls, etc)
> and will issue commands like bridge the calls to their destination, etc.
>
> Any help would be appreciated!
>
> Thanks
>
> David
>
> On Fri, Aug 3, 2012 at 1:13 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Any thoughts?
>>
>>
>> On Thu, Aug 2, 2012 at 9:21 PM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Actually no. I'm writing a script to control FS, incoming calls will be
>>> screened and routing will de done via this script... I already have this in
>>> Lua, but everyting a call comes in a new script is lunched, and I don't
>>> want that anymore. So I'm trying to write an ESL script wich will connect
>>> to FS via ESL and receive events, act accordingly and instruct fs what to
>>> do, in terms of routing, channel variables definition (for later use in
>>> xml_curl_cdr, etc.
>>>
>>> Maybe my approach is not the correct one...
>>>
>>> Thanks for your help...
>>>
>>> David
>>>
>>> On Thu, Aug 2, 2012 at 9:07 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Well, it all depends on how you're calling this script. Is it being
>>>> called from the dialplan?
>>>> -MC
>>>>
>>>>
>>>> On Thu, Aug 2, 2012 at 11:25 AM, David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>>> Ouch!
>>>>>
>>>>> Ok, this is not like in Lua... Ok, I'll try that.... is there any
>>>>> documentation regarding what can and can't be set and the conditions for
>>>>> setting them? What about bridging channels, etc?
>>>>>
>>>>> :(
>>>>>
>>>>> David
>>>>>
>>>>> On Thu, Aug 2, 2012 at 8:19 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> Can't use execute and set unless this is an outbound ESL connection,
>>>>>> which it seems not to be. I'd try this:
>>>>>>
>>>>>> $con->api("uuid_setvar","$uuid customer_company $local_variable");
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> On Thu, Aug 2, 2012 at 11:01 AM, David Villasmil <
>>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>>
>>>>>>> Hello Michael,
>>>>>>>
>>>>>>> Yes, there's a call in progress. Executing it like you said worked.
>>>>>>> (api)
>>>>>>>
>>>>>>>
>>>>>>> Now, I'm trying to set a variable like so:
>>>>>>>
>>>>>>> $con->execute("set","customer_company             = " .
>>>>>>>  $local_variable , $uuid );
>>>>>>>
>>>>>>> But it doesn't get set...
>>>>>>>
>>>>>>> :(
>>>>>>>
>>>>>>> Thanks for your help!
>>>>>>>
>>>>>>> David
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Aug 2, 2012 at 6:25 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>>
>>>>>>>> David,
>>>>>>>>
>>>>>>>> Do you have a call in progress at this point? If not then you'll
>>>>>>>> need to supply a uuid of a live call, as mentioned here:
>>>>>>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library#execute
>>>>>>>>
>>>>>>>> Remember, "execute" means "execute dialplan application" so if
>>>>>>>> there's no channel then "execute" really doesn't mean a whole lot.
>>>>>>>>
>>>>>>>> Alternatively you could try something like this:
>>>>>>>>
>>>>>>>> $con->api("log","WARNING Don't cross the streams!!");
>>>>>>>>
>>>>>>>> Remember this rule of thumb: pretty much anything you type at
>>>>>>>> fs_cli is an "API" and therefore you can use $con->api(), whereas anything
>>>>>>>> that is a diaplan application requires an actual channel on which to run
>>>>>>>> $con->execute().
>>>>>>>>
>>>>>>>> Hope that makes sense... :)
>>>>>>>>
>>>>>>>> -Michael
>>>>>>>>
>>>>>>>> On Thu, Aug 2, 2012 at 8:43 AM, David Villasmil <
>>>>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Hello Guys,
>>>>>>>>>
>>>>>>>>> I'm starting off with ESL, which is cool, but I'm trying to log to
>>>>>>>>> the console like Lua's "freeswitch.consoleLog("info", "BLAH BLAH");" doing
>>>>>>>>> it like:
>>>>>>>>>
>>>>>>>>> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>>>>>>>>> $con->execute("log", "1, BlahBlah");
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> But nothing gets in the log files or console... and I can't find
>>>>>>>>> any documentation as to how to log using "execute"...
>>>>>>>>>
>>>>>>>>> any ideas?
>>>>>>>>>
>>>>>>>>> Thanks!
>>>>>>>>>
>>>>>>>>> David
>>>>>>>>>
>>>>>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From james.bravo at redmatter.com  Fri Aug  3 16:54:17 2012
From: james.bravo at redmatter.com (James Bravo)
Date: Fri, 3 Aug 2012 13:54:17 +0100
Subject: [Freeswitch-users] CUSTOM event messages missing FreeSWITCH-Hostname
Message-ID: <501BC9F9.3070703@redmatter.com>

Hi

We've recently upgraded from version 1.0.head to 1.2.0-rc2
and noticed that certain events sent from php code using
the ESP library have certain core headers missing, in
particular 'FreeSWITCH-Hostname'.
For example, using '/events plain all' in fs_cli,
version 1.0.head would show

     RECV EVENT
     Event-Subclass: rmCTI::TRANSFER_ANSWER
     Core-UUID: 47527f1f-b6c3-4f74-9f16-32f565ee4123
     FreeSWITCH-Hostname: s01pbx01.test.com
     FreeSWITCH-Switchname: s01pbx01.test.com
     FreeSWITCH-IPv4: 211.141.141.131
     FreeSWITCH-IPv6: ::1
     Event-Date-Local: 2012-08-03 10:43:06
     Event-Date-GMT: Fri, 03 Aug 2012 09:43:06 GMT
     Event-Date-Timestamp: 1343986986238409
     Event-Calling-File: mod_event_socket.c
     Event-Calling-Function: read_packet
     Event-Calling-Line-Number: 1188
     Command: sendevent CUSTOM
     Event-Name: CUSTOM
     Channel-Name: sofia/internal1/1202 at sip.test.com
     rmCTI-UUID: aebc98dc-22da-4ad1-8234-912487329123
     rmCTI-Other-UUID: b24a8783-16b1-495e-9c4e-5ea52c362123
     rmCTI-Info: Transfer to new b-leg answered

where as version 1.2.0-rc2 it shows

     RECV EVENT
     Event-Subclass: rmCTI::TRANSFER_ANSWER
     Command: sendevent CUSTOM
     Event-Name: CUSTOM
     Channel-Name: sofia/internal1/1202 at sip.test.com
     rmCTI-UUID: c3c7b331-0b8b-4ea5-834f-9d72fd446123
     rmCTI-Other-UUID: db75c60b-a258-4b47-8d51-a1b7ac86f123
     rmCTI-Info: Transfer to new b-leg answered
     Event-UUID: f47597b2-9e8c-4508-be57-3e1eed35c123

Is there any reason these core headers are missing?
Is it something to do with CUSTOM events?

Thanks in advance
James Bravo
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From m.prevail at gmail.com  Fri Aug  3 18:22:29 2012
From: m.prevail at gmail.com (Prevail Magid)
Date: Fri, 3 Aug 2012 17:22:29 +0300
Subject: [Freeswitch-users] How Bridge users from one node of FS to
	conference in other node.
Message-ID: <CANJXrA9Bb9CTxVapd782QHL8J0_BD6a3RKvx+iGq98YoYJGmmg@mail.gmail.com>

Hi,
 I try to create FreeSWITCH HA Clustering.
 I have node A and node B with ODBC and ESL client.

Node A have conference. At Node B calling users which should be connected
to conference on node A.

As I read here :
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-January/067384.htmlneed
just bridge a user to conference.

But how? If I use command  like uuid_bridge <caller_uuid> <conference_uuid>
I just disconnected.
How Node B can know about conference on node A? As I understand FS  do not
store this information to table that create by ODBC?
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From msc at freeswitch.org  Fri Aug  3 20:03:13 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Aug 2012 09:03:13 -0700
Subject: [Freeswitch-users] External profile & NAT
In-Reply-To: <501BA7B7.1080709@thewinelake.com>
References: <4FDF35D1.306@the800group.com>
	<CAKzWOxUW5ehDU+5cc=FCLnXX6=_iA0Gr0REwa0=wz9huztAExw@mail.gmail.com>
	<501BA7B7.1080709@thewinelake.com>
Message-ID: <CAKzWOxUS0EUiDNfqTVmpHr5kNx29+D81TtBVWKC6_i7gV1N0Ag@mail.gmail.com>

On Fri, Aug 3, 2012 at 3:28 AM, Alex <alex at thewinelake.com> wrote:

>  Does reload mod_sofia harm any calls in progress?
>

It will only harm every single SIP call in progress. ;)

-MC
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From gustavomarsico at gmail.com  Fri Aug  3 20:30:21 2012
From: gustavomarsico at gmail.com (ServTelar)
Date: Fri, 3 Aug 2012 13:30:21 -0300
Subject: [Freeswitch-users] Directory xml_curl, gets Can't find user
In-Reply-To: <3B0CFE1B-2F8F-428B-85DB-B420A2C5FE18@gmail.com>
References: <3B0CFE1B-2F8F-428B-85DB-B420A2C5FE18@gmail.com>
Message-ID: <C24B229D-99BB-4F33-8E0C-C17FA378A029@gmail.com>

I tried with 

sofia/internal/2692128%10.8.0.70

Is this the way that i should work instead of using user/ ?



On Aug 3, 2012, at 11:39 AM, ServTelar wrote:

> Hi all
> 
> It seems this is a very old one, but certainly I didn't find any solution to this.
> 
> Using XML for directory everything works fine. But if I try using xml_curl, sending response:
> 
> <document type="freeswitch/xml">
>  <section description="FreeSWITCH Directory" name="directory">
>    <domain name="10.8.0.70">
>      <params>
>        <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>      </params>
>      <groups>
>        <group name="default">
>          <users>
>            <user id="2692128">
>              <params>
>                <param name="a1-hash" value="7d73f412ca332fedd81a297b64c0b271"/>
>                <param name="vm-password" value="2128"/>
>              </params>
>              <variables>
>                <variable name="accountcode" value="2692128"/>
>                <variable name="user_context" value="default"/>
>                <variable name="effective_caller_id_name" value="2692128"/>
>                <variable name="effective_caller_id_number" value="2692128"/>
>                <variable name="outbound_caller_id_name" value="2692128"/>
>                <variable name="outbound_caller_id_number" value="2692128"/>
>              </variables>
>            </user>
>          </users>
>        </group>
>      </groups>
>    </domain>
>  </section>
> </document>
> 
> Registration is accepted:
> 
> Registrations:
> =================================================================================================
> Call-ID:    	939540755
> User:       	2692128 at 10.8.0.70
> Contact:    	2692128 <sip:2692128 at 10.0.0.10:5060;line=3dfb73927e26966;fs_nat=yes;fs_path=sip%3A2692128%4010.94.56.6%3A5060%3Bline%3D3dfb73927e26966>
> Agent:      	iSoftPhone 3.4110
> Status:     	Registered(UDP-NAT)(unknown) EXP(2012-08-03 11:35:30) EXPSECS(1688)
> Host:       	demo
> IP:         	10.94.56.6
> Port:       	5060
> Auth-User:  	2692128
> Auth-Realm: 	10.8.0.70
> MWI-Account:	2692128 at 10.8.0.70
> 
> 
> But if I made a call pointing to that user I just got:
> 
> 2012-08-03 11:05:38.261385 [WARNING] mod_dptools.c:3294 Can't find user [2692128 at 10.8.0.70]
> 2012-08-03 11:05:38.281377 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
> 2012-08-03 11:05:38.281377 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT]
> 
> Domain are set both to the IP:
> 
>  <X-PRE-PROCESS cmd="set" data="domain=10.8.0.70"/>
>  <X-PRE-PROCESS cmd="set" data="domain_name=10.8.0.70"/>
> 
> And matches with the internal
> 
> freeswitch at internal> sofia status 
>                     Name	   Type	                                      Data	State
> =================================================================================================
>                10.8.0.70	  alias	                                  internal	ALIASED
>                 internal	profile	              sip:mod_sofia at 10.8.0.70:5060	RUNNING (0)
> 
> 
> Can someone drop me a line to point where I should look to?
> 
> 
> Thanks in advance
> 
> Gustavo
> 
> 



From mario_fs at mgtech.com  Fri Aug  3 20:30:56 2012
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 3 Aug 2012 09:30:56 -0700
Subject: [Freeswitch-users] Anyine have Bria iOS tipes for FreeSwitch?
In-Reply-To: <CC402B6B.3ACBF%krice@freeswitch.org>
References: <CC402B6B.3ACBF%krice@freeswitch.org>
Message-ID: <DD08729F-897B-400C-995E-6333A2CDA9DB@mgtech.com>

Thanks for asking, and hope anyone can shed light on the issues, been working for about , bottom line is calls don't get answered and they say FS is either sending cancel or bye, but why when all the other extensions keep ringing and I can answer the call using them. I am also trying to get the bridge command to isolate what is sent to Bria to make sure variables are not causing issues but since I am using call_group(s) that is not possible and enterprise won't work. It's a pain because these are intermittent problems (except for #5) so it's impossible to isolate so far:

PROBLEM 1: This is not a consistent issue, I opened the cover and did nothing else and the call ended. The other phones kept ringing so I could answer. I am having a lot of Bria problems even after deleting the App and starting all over again.
Counterpath: This time, we got a "Cancel" instead of a "BYE" from the other end. This is consistent with the far end ending the call before the iPad answers. Bria definitely gets the "hangup" signal. I'd recommend checking the server logs. I can't speculate as to why the server sends the BYE.

PROBLEM 2: An incoming call rang the iPad, But when I touched the notification Bria stopped ringing and ended the call without answering it (the PBX still ringing other phones so Bria never picked up).  I have been having issues with not being able to answer like this and would appreciate if that can be looked at in the trace.
Counterpath: The first call was ended by the remote party. As soon as we sent the 200 OK to answer, the server sends an ACK (to acknowledge receipt of the 200), then it sends a BYE to hang up. Bria hangs up accordingly.

PROBLEM 3: This is not a consistent issue, it normally works fine almost all the time. I was using Safari on the New iPad when a call came in to the system, there was no notification on Bria and no ringing on Bria so I could not answer. When I looked at the PBX log I found Bria sent BUSY but I was not on a call. Bria was an active app but not set to run in the background. Problem occurred on the New iPad, OK on iPad 2 but don't know if difference has anything to do with it.

PROBLEM 4: Intermittently, when we make an outbound outside call, there is no sound, no ringing, etc. but the call is made. Retrying the call will stay this way until Bria is stopped and restarted then the same call works. This is an intermittent problem but happens about 20 percent of the time. I do not think this is a NATting issue since there is no NAT involved in the local network. This problem does not occur with other devices, only Bria.

5. Bria only rings for 25 seconds if the cover is closed or not Bria not being displayed: After many traces Counterpath points the finger at Apple, says the notification has limits on how long it can ring. This one is not a FS problem, added in case someone else has the same issue.


BRIA: Version 2.1.2 build 12566

DEVICE: iPad 3, Bria using TCP, with background OFF.

ENVIRONMENT:  iPad on an internal network connected to a FreeSwitch PBX running on a Mac Mini, there is NO NATing involved since the router handles it.

What I tried:
Global IP Wifi ON/OFF- currently OFF.

SETTINGS:
Notification for Bria is On, Badge app icon, sounds, view in lock screen

BRIA Prefererences: Run in background off, all other ON except forward calls

Advanced settings: VPN off, VAD off, NR TX ON, NR RX OFF, QOS ON, RTP Port 4000, Codecs defaulted, Verify TLS OFF (not using TLS), Sip Misc both ON

Account: Enabled, Display as caller ID, username 110, PWD, Domain, VM number 4001,

Account Advanced: Account additional defauls, Send DTMF RFC2833, Network Traversal both OFF, Call dialing OFF, Transport/Sec TCP/NEVER, Incoming calls ON, refresh interval 900, Keep alive 30, Single register OFF, Passive time OFF, Connection reuse OFF, Enable IMS OFF


On Aug 2, 2012, at 11:04 AM, Ken Rice wrote:

> What kind of problems are you having? I use Bria on my iPad all the time and
> don't seem to have issues with it
> 
> 
> On 8/2/12 12:05 PM, "Mario G" <mario_fs at mgtech.com> wrote:
> 
>> I am at my wits end after several months of Bria problems. Counterpath keeps
>> pointing the finger at FreeSwitch. I am wondering if anyone here has Bria
>> working well and if they can share tips user definitions? Bria is set for TCP
>> to reduce battery usage. Thanks.
>> Mario G
>> 



From mario_fs at mgtech.com  Fri Aug  3 20:34:18 2012
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 3 Aug 2012 09:34:18 -0700
Subject: [Freeswitch-users] How to file FreeSwitch Enhancements requests?
In-Reply-To: <CAKzWOxVrVn_cQ1brYhvF7g4ZYdqq3PtsRLJ+37YWN65wKPQ1QA@mail.gmail.com>
References: <D6CCBDDC-EC0A-46FC-92F1-35177A3BDB5B@mgtech.com>
	<CAKzWOxVrVn_cQ1brYhvF7g4ZYdqq3PtsRLJ+37YWN65wKPQ1QA@mail.gmail.com>
Message-ID: <1C821F6C-85ED-47F8-B5D0-65C37A95CAE0@mgtech.com>

WIll do, I guess I can code each user separately for now to solve the problems....

On Aug 2, 2012, at 10:05 AM, Michael Collins wrote:

> You can do this with a Jira. Also, after you open the Jira go ahead and put up a bounty on the wiki:
> http://wiki.freeswitch.org/wiki/Bounty
> 
> Sometimes that will grease the wheels.
> 
> -Michael
> 
> On Thu, Aug 2, 2012 at 9:50 AM, Mario G <mario_fs at mgtech.com> wrote:
> Is there an official way to suggest FreeSwitch enhancements? Where should it go and how much doc/explanation is needed? In this case:
> 
> 1. in bridge: Add the ability to specify variables that apply only to one group_call but all users in that group.
> 
> 2. In bridge: Add the ability to allow leg_delay_start, etc. to work using enterprise syntax.
> 
> I posted this yesterday which may explain more: (has gotten no response)
> 
> 
> What I need: to have different variable/settings for each target group in a bridge command, and allow leg_delay_start to work. It's related to trying to fix other issues and holding up progress for them. Thanks for any help,
> Mario G
> 
> I scoured the wiki and tried many things. Normally, enterprise syntax would solve this, but there is a catch: you can't use leg_delay_start which I need, it is ignored when using enterprise. I found that using brackets [] only applies the variable to the first user in the group, not the whole group so that won't work.
> 
> What I have now but need to have alert_info moved/apply to the mgt group only: 
> <action application="bridge" data="{originate_timeout=45,alert_info=n=${lua_ringtone}}${group_call(mgtbria@${domain_name}+A)},${group_call(mgt@${domain_name}+A)},[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551212,[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551313"/>
>     
> 
>  This solves variables for groups but breaks leg_delay_start, etc.:
> <action application="bridge" data="<originate_timeout=45>}${group_call(mgtbria@${domain_name}+A)}:_:{alert_info=n=${lua_ringtone}${group_call(mgt@${domain_name}+A)}:_:[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551212,[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551313"/> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From robert.hadley at teotech.com  Fri Aug  3 20:59:16 2012
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 3 Aug 2012 16:59:16 +0000
Subject: [Freeswitch-users] Help so inbound-late-negotiation=true and SRTP
 can coexist in same profile?
Message-ID: <71943DD5C22943448A24B7C5CDC23807108D17BE@CH1PRD0411MB430.namprd04.prod.outlook.com>

Hi FS users,

We want to use <param name="inbound-late-negotiation" value="true"/> in the internal sip profile, and <action application="inherit_codec=true"/> in the dialplan so as to minimize codec transcoding in Freeswitch.  But we also have other devices registered to the same sip profile using TLS/SRTP.  With this combination of settings SRTP is broken on the B-leg, resulting in RTP media.  Is there a way to override the sip profile parameter inbound-late-negotiation on a per call/channel basis?

Freeswitch's default internal sip_profile looks like it should support both UDP (5060) and TLS/SRTP (5061), but the above limitation makes the internal sip profile unusable for TLS/SRTP.  Is there any way for SRTP and late negotiation to coexist in the same profile?

Thanks,
Robert

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From philq at qsystemsengineering.com  Fri Aug  3 22:02:41 2012
From: philq at qsystemsengineering.com (Phil Quesinberry)
Date: Fri, 03 Aug 2012 14:02:41 -0400
Subject: [Freeswitch-users] G711-16?
Message-ID: <026201cd71a2$2fc172a0$8f4457e0$@com>

Does FS support G711/PCMU 16K?  If so, how do you specify it in the codec
list?  I can't seem to find any info on it and 'show codecs' doesn't show it
so my first assumption is that it doesn't.  Since Aastras and other
endpoints support it, I thought I'd better ask before assuming that was the
case.

 

Thanks,

 

Phil Quesinberry

Q Systems Engineering, Inc.

Electronic Controls and Embedded Systems Development

(410) 969-8002

http://www.qsystemsengineering.com

 

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From william.king at quentustech.com  Fri Aug  3 22:05:21 2012
From: william.king at quentustech.com (William King)
Date: Fri, 03 Aug 2012 14:05:21 -0400
Subject: [Freeswitch-users] Directory xml_curl, gets Can't find user
In-Reply-To: <C24B229D-99BB-4F33-8E0C-C17FA378A029@gmail.com>
References: <3B0CFE1B-2F8F-428B-85DB-B420A2C5FE18@gmail.com>
	<C24B229D-99BB-4F33-8E0C-C17FA378A029@gmail.com>
Message-ID: <501C12E1.7040809@quentustech.com>

sofia/internal/2692128 at 10.8.0.70
or
user/2692128

Should work.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 08/03/2012 12:30 PM, ServTelar wrote:
> I tried with
>
> sofia/internal/2692128%10.8.0.70
>
> Is this the way that i should work instead of using user/ ?
>
>
>
> On Aug 3, 2012, at 11:39 AM, ServTelar wrote:
>
>> Hi all
>>
>> It seems this is a very old one, but certainly I didn't find any solution to this.
>>
>> Using XML for directory everything works fine. But if I try using xml_curl, sending response:
>>
>> <document type="freeswitch/xml">
>>   <section description="FreeSWITCH Directory" name="directory">
>>     <domain name="10.8.0.70">
>>       <params>
>>         <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>       </params>
>>       <groups>
>>         <group name="default">
>>           <users>
>>             <user id="2692128">
>>               <params>
>>                 <param name="a1-hash" value="7d73f412ca332fedd81a297b64c0b271"/>
>>                 <param name="vm-password" value="2128"/>
>>               </params>
>>               <variables>
>>                 <variable name="accountcode" value="2692128"/>
>>                 <variable name="user_context" value="default"/>
>>                 <variable name="effective_caller_id_name" value="2692128"/>
>>                 <variable name="effective_caller_id_number" value="2692128"/>
>>                 <variable name="outbound_caller_id_name" value="2692128"/>
>>                 <variable name="outbound_caller_id_number" value="2692128"/>
>>               </variables>
>>             </user>
>>           </users>
>>         </group>
>>       </groups>
>>     </domain>
>>   </section>
>> </document>
>>
>> Registration is accepted:
>>
>> Registrations:
>> =================================================================================================
>> Call-ID:    	939540755
>> User:       	2692128 at 10.8.0.70
>> Contact:    	2692128<sip:2692128 at 10.0.0.10:5060;line=3dfb73927e26966;fs_nat=yes;fs_path=sip%3A2692128%4010.94.56.6%3A5060%3Bline%3D3dfb73927e26966>
>> Agent:      	iSoftPhone 3.4110
>> Status:     	Registered(UDP-NAT)(unknown) EXP(2012-08-03 11:35:30) EXPSECS(1688)
>> Host:       	demo
>> IP:         	10.94.56.6
>> Port:       	5060
>> Auth-User:  	2692128
>> Auth-Realm: 	10.8.0.70
>> MWI-Account:	2692128 at 10.8.0.70
>>
>>
>> But if I made a call pointing to that user I just got:
>>
>> 2012-08-03 11:05:38.261385 [WARNING] mod_dptools.c:3294 Can't find user [2692128 at 10.8.0.70]
>> 2012-08-03 11:05:38.281377 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
>> 2012-08-03 11:05:38.281377 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT]
>>
>> Domain are set both to the IP:
>>
>>   <X-PRE-PROCESS cmd="set" data="domain=10.8.0.70"/>
>>   <X-PRE-PROCESS cmd="set" data="domain_name=10.8.0.70"/>
>>
>> And matches with the internal
>>
>> freeswitch at internal>  sofia status
>>                      Name	   Type	                                      Data	State
>> =================================================================================================
>>                 10.8.0.70	  alias	                                  internal	ALIASED
>>                  internal	profile	              sip:mod_sofia at 10.8.0.70:5060	RUNNING (0)
>>
>>
>> Can someone drop me a line to point where I should look to?
>>
>>
>> Thanks in advance
>>
>> Gustavo
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From curriegrad2004 at gmail.com  Fri Aug  3 22:16:30 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 3 Aug 2012 11:16:30 -0700
Subject: [Freeswitch-users] G711-16?
In-Reply-To: <026201cd71a2$2fc172a0$8f4457e0$@com>
References: <026201cd71a2$2fc172a0$8f4457e0$@com>
Message-ID: <CAMWiMy9VEVo25PoexRE9Ob9ghYS7E19bBWqg+KE0p1fmAvxMcw@mail.gmail.com>

Unofficially yes if you proxy the media once you see whatever SDP is
for G711-16K

On Fri, Aug 3, 2012 at 11:02 AM, Phil Quesinberry
<philq at qsystemsengineering.com> wrote:
> Does FS support G711/PCMU 16K?  If so, how do you specify it in the codec
> list?  I can?t seem to find any info on it and ?show codecs? doesn?t show it
> so my first assumption is that it doesn?t.  Since Aastras and other
> endpoints support it, I thought I?d better ask before assuming that was the
> case.
>
>
>
> Thanks,
>
>
>
> Phil Quesinberry
>
> Q Systems Engineering, Inc.
>
> Electronic Controls and Embedded Systems Development
>
> (410) 969-8002
>
> http://www.qsystemsengineering.com
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From all.eforums at gmail.com  Fri Aug  3 23:44:07 2012
From: all.eforums at gmail.com (A E G)
Date: Fri, 3 Aug 2012 15:44:07 -0400
Subject: [Freeswitch-users] SOLVED: Asterisk don't like multiple m=
 lines in the SDP [Was: Error: 88 Incompatible Destination after upgrade]
In-Reply-To: <CADzY+JiUnUY+U=jrnVzBWNgYqp42jnEksHGA-DWfMnbt-Use9A@mail.gmail.com>
References: <CADzY+JiUnUY+U=jrnVzBWNgYqp42jnEksHGA-DWfMnbt-Use9A@mail.gmail.com>
Message-ID: <CADzY+JiUj=ExwEVL5hS0H=0uaNB9LjVoXXju0-mGejiTukK0ww@mail.gmail.com>

...and I'm not done yet.

So while this one below solved the problem, how do we manage this so I
don't always suppress all the m= lines in my SDP when sending calls to all
the external gateways, but do that only when sending calls to any system
that doesn't like it?

As I said before, I tried doing this just before the bridge, but didn't
work. Only seems to work as a PRE-PROCESS Global setting.

Thx

On Fri, Aug 3, 2012 at 1:44 AM, A E G <all.eforums at gmail.com> wrote:

> Well turns out learning to use Google better always helps.
>
> Found a nugget of wisdom that <action application="set"
> data="sdp_m_per_ptime=false"/> should've actually been in the vars.xml
>
> Seems to have fixed it.
>
>
>
> On Fri, Aug 3, 2012 at 12:01 AM, A E G <all.eforums at gmail.com> wrote:
>
>> Gents,
>>
>> running version: 1.2.0-rc2+git~20120731T213556Z~e97da8e20a
>>
>> have been fighting with this for an hour or more...have done a bit of
>> research on the list and Google itself, scoured the Wiki etc. but can't
>> seem to figure out where to set the codecs to be "well recd" by the remote
>> SIP peer. This used to work fine until I upgraded 2 days ago and it stopped
>> working without my doing / changing anything.
>>
>> The remote SIP peer is {*} which doesn't like multiple m= lines with
>> differing ptime values.
>>
>> Debug on near-end says: "Originate Resulted in Error Cause: 88
>> [INCOMPATIBLE_DESTINATION]"
>>
>> Full SDP here:
>>
>> v=0
>> o=FreeSWITCH 1343934933 1343934934 IN IP4 192.168.1.80
>> s=FreeSWITCH
>> c=IN IP4 192.168.1.80
>> t=0 0
>> m=audio 28884 RTP/AVP 98 8 3 101 13
>> a=rtpmap:98 L16/16000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=sendrecv
>> m=audio 28884 RTP/AVP 0 101 13
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:40
>> a=sendrecv
>>
>> Far end says:  Rejecting non-primary audio stream: audio 28884 RTP/AVP 0
>> 101 13
>>
>> I have tried to play around with the codec globals in vars.xml, to no
>> avail.
>>
>> have also added stuff directly in the dialplan like so:
>>
>> <action application="set" data="hangup_after_bridge=true"/>
>> <action application="set" data="absolute_codec_string=PCMU at 40i"/>
>> <action application="set" data="sdp_m_per_ptime=false"/>
>> <action application="bridge" data="sofia/gateway/mycore1/1111"/>
>>
>>
>> but no dice.
>>
>> Tried to remove the "absolute_codec_string", still no dice.
>>
>> I read about the "sdp_m_per_ptime"  that lets me "cheat" but that ain't
>> doing anything either.
>>
>> What gives?
>>
>> Thx in advance
>>
>>
>>
>>
>>
>>
>
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From msc at freeswitch.org  Fri Aug  3 23:51:51 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Aug 2012 12:51:51 -0700
Subject: [Freeswitch-users] SOLVED: Asterisk don't like multiple m=
 lines in the SDP [Was: Error: 88 Incompatible Destination after upgrade]
In-Reply-To: <CADzY+JiUj=ExwEVL5hS0H=0uaNB9LjVoXXju0-mGejiTukK0ww@mail.gmail.com>
References: <CADzY+JiUnUY+U=jrnVzBWNgYqp42jnEksHGA-DWfMnbt-Use9A@mail.gmail.com>
	<CADzY+JiUj=ExwEVL5hS0H=0uaNB9LjVoXXju0-mGejiTukK0ww@mail.gmail.com>
Message-ID: <CAKzWOxVBbrTj1nWbqiT7=5d+LFUj6mkBTbKhieM7M+YHbW3CZw@mail.gmail.com>

Did you solve it with setting a channel variable or a Sofia profile
setting? If a channel variable then you should be able to do a little
dialplan logic and set the value only if certain conditions are met, i.e.
if a certain gateway is going to be used.

-MC

On Fri, Aug 3, 2012 at 12:44 PM, A E G <all.eforums at gmail.com> wrote:

> ...and I'm not done yet.
>
> So while this one below solved the problem, how do we manage this so I
> don't always suppress all the m= lines in my SDP when sending calls to all
> the external gateways, but do that only when sending calls to any system
> that doesn't like it?
>
> As I said before, I tried doing this just before the bridge, but didn't
> work. Only seems to work as a PRE-PROCESS Global setting.
>
> Thx
>
> On Fri, Aug 3, 2012 at 1:44 AM, A E G <all.eforums at gmail.com> wrote:
>
>> Well turns out learning to use Google better always helps.
>>
>> Found a nugget of wisdom that <action application="set"
>> data="sdp_m_per_ptime=false"/> should've actually been in the vars.xml
>>
>> Seems to have fixed it.
>>
>>
>>
>> On Fri, Aug 3, 2012 at 12:01 AM, A E G <all.eforums at gmail.com> wrote:
>>
>>> Gents,
>>>
>>> running version: 1.2.0-rc2+git~20120731T213556Z~e97da8e20a
>>>
>>> have been fighting with this for an hour or more...have done a bit of
>>> research on the list and Google itself, scoured the Wiki etc. but can't
>>> seem to figure out where to set the codecs to be "well recd" by the remote
>>> SIP peer. This used to work fine until I upgraded 2 days ago and it stopped
>>> working without my doing / changing anything.
>>>
>>> The remote SIP peer is {*} which doesn't like multiple m= lines with
>>> differing ptime values.
>>>
>>> Debug on near-end says: "Originate Resulted in Error Cause: 88
>>> [INCOMPATIBLE_DESTINATION]"
>>>
>>> Full SDP here:
>>>
>>> v=0
>>> o=FreeSWITCH 1343934933 1343934934 IN IP4 192.168.1.80
>>> s=FreeSWITCH
>>> c=IN IP4 192.168.1.80
>>> t=0 0
>>> m=audio 28884 RTP/AVP 98 8 3 101 13
>>> a=rtpmap:98 L16/16000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>> a=sendrecv
>>> m=audio 28884 RTP/AVP 0 101 13
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:40
>>> a=sendrecv
>>>
>>> Far end says:  Rejecting non-primary audio stream: audio 28884 RTP/AVP
>>> 0 101 13
>>>
>>> I have tried to play around with the codec globals in vars.xml, to no
>>> avail.
>>>
>>> have also added stuff directly in the dialplan like so:
>>>
>>> <action application="set" data="hangup_after_bridge=true"/>
>>> <action application="set" data="absolute_codec_string=PCMU at 40i"/>
>>> <action application="set" data="sdp_m_per_ptime=false"/>
>>> <action application="bridge" data="sofia/gateway/mycore1/1111"/>
>>>
>>>
>>> but no dice.
>>>
>>> Tried to remove the "absolute_codec_string", still no dice.
>>>
>>> I read about the "sdp_m_per_ptime"  that lets me "cheat" but that ain't
>>> doing anything either.
>>>
>>> What gives?
>>>
>>> Thx in advance
>>>
>>>
>>>
>>>
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From all.eforums at gmail.com  Sat Aug  4 00:03:45 2012
From: all.eforums at gmail.com (A E G)
Date: Fri, 3 Aug 2012 16:03:45 -0400
Subject: [Freeswitch-users] SOLVED: Asterisk don't like multiple m=
 lines in the SDP [Was: Error: 88 Incompatible Destination after upgrade]
In-Reply-To: <CAKzWOxVBbrTj1nWbqiT7=5d+LFUj6mkBTbKhieM7M+YHbW3CZw@mail.gmail.com>
References: <CADzY+JiUnUY+U=jrnVzBWNgYqp42jnEksHGA-DWfMnbt-Use9A@mail.gmail.com>
	<CADzY+JiUj=ExwEVL5hS0H=0uaNB9LjVoXXju0-mGejiTukK0ww@mail.gmail.com>
	<CAKzWOxVBbrTj1nWbqiT7=5d+LFUj6mkBTbKhieM7M+YHbW3CZw@mail.gmail.com>
Message-ID: <CADzY+JiUDcq35rX2Y55JL_VwOnZ9ckKvrVNa50AudE7qOwqkWQ@mail.gmail.com>

On Fri, Aug 3, 2012 at 3:51 PM, Michael Collins <msc at freeswitch.org> wrote:

> Did you solve it with setting a channel variable or a Sofia profile
> setting? If a channel variable then you should be able to do a little
> dialplan logic and set the value only if certain conditions are met, i.e.
> if a certain gateway is going to be used.
>
> -MC
>
>
Solved it using the Global variable in vars.xml. Sticking
sdp_m_per_ptime=false as a channel variable (just before calling bridge)
didn't seem to have any affect on the behaviour. Didn't try it in the
specific profile to which that gateway belongs.


> On Fri, Aug 3, 2012 at 12:44 PM, A E G <all.eforums at gmail.com> wrote:
>
>> ...and I'm not done yet.
>>
>> So while this one below solved the problem, how do we manage this so I
>> don't always suppress all the m= lines in my SDP when sending calls to all
>> the external gateways, but do that only when sending calls to any system
>> that doesn't like it?
>>
>> As I said before, I tried doing this just before the bridge, but didn't
>> work. Only seems to work as a PRE-PROCESS Global setting.
>>
>> Thx
>>
>> On Fri, Aug 3, 2012 at 1:44 AM, A E G <all.eforums at gmail.com> wrote:
>>
>>> Well turns out learning to use Google better always helps.
>>>
>>> Found a nugget of wisdom that <action application="set"
>>> data="sdp_m_per_ptime=false"/> should've actually been in the vars.xml
>>>
>>> Seems to have fixed it.
>>>
>>>
>>>
>>> On Fri, Aug 3, 2012 at 12:01 AM, A E G <all.eforums at gmail.com> wrote:
>>>
>>>> Gents,
>>>>
>>>> running version: 1.2.0-rc2+git~20120731T213556Z~e97da8e20a
>>>>
>>>> have been fighting with this for an hour or more...have done a bit of
>>>> research on the list and Google itself, scoured the Wiki etc. but can't
>>>> seem to figure out where to set the codecs to be "well recd" by the remote
>>>> SIP peer. This used to work fine until I upgraded 2 days ago and it stopped
>>>> working without my doing / changing anything.
>>>>
>>>> The remote SIP peer is {*} which doesn't like multiple m= lines with
>>>> differing ptime values.
>>>>
>>>> Debug on near-end says: "Originate Resulted in Error Cause: 88
>>>> [INCOMPATIBLE_DESTINATION]"
>>>>
>>>> Full SDP here:
>>>>
>>>> v=0
>>>> o=FreeSWITCH 1343934933 1343934934 IN IP4 192.168.1.80
>>>> s=FreeSWITCH
>>>> c=IN IP4 192.168.1.80
>>>> t=0 0
>>>> m=audio 28884 RTP/AVP 98 8 3 101 13
>>>> a=rtpmap:98 L16/16000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>> a=sendrecv
>>>> m=audio 28884 RTP/AVP 0 101 13
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:40
>>>> a=sendrecv
>>>>
>>>> Far end says:  Rejecting non-primary audio stream: audio 28884 RTP/AVP
>>>> 0 101 13
>>>>
>>>> I have tried to play around with the codec globals in vars.xml, to no
>>>> avail.
>>>>
>>>> have also added stuff directly in the dialplan like so:
>>>>
>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>> <action application="set" data="absolute_codec_string=PCMU at 40i"/>
>>>> <action application="set" data="sdp_m_per_ptime=false"/>
>>>> <action application="bridge" data="sofia/gateway/mycore1/1111"/>
>>>>
>>>>
>>>> but no dice.
>>>>
>>>> Tried to remove the "absolute_codec_string", still no dice.
>>>>
>>>> I read about the "sdp_m_per_ptime"  that lets me "cheat" but that ain't
>>>> doing anything either.
>>>>
>>>> What gives?
>>>>
>>>> Thx in advance
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Aug  4 00:26:10 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Aug 2012 13:26:10 -0700
Subject: [Freeswitch-users] SOLVED: Asterisk don't like multiple m=
 lines in the SDP [Was: Error: 88 Incompatible Destination after upgrade]
In-Reply-To: <CADzY+JiUDcq35rX2Y55JL_VwOnZ9ckKvrVNa50AudE7qOwqkWQ@mail.gmail.com>
References: <CADzY+JiUnUY+U=jrnVzBWNgYqp42jnEksHGA-DWfMnbt-Use9A@mail.gmail.com>
	<CADzY+JiUj=ExwEVL5hS0H=0uaNB9LjVoXXju0-mGejiTukK0ww@mail.gmail.com>
	<CAKzWOxVBbrTj1nWbqiT7=5d+LFUj6mkBTbKhieM7M+YHbW3CZw@mail.gmail.com>
	<CADzY+JiUDcq35rX2Y55JL_VwOnZ9ckKvrVNa50AudE7qOwqkWQ@mail.gmail.com>
Message-ID: <CAKzWOxU6PYAjafbef++inPFSU5Oc8VcK9=KE5rEwPQPhSti2Zg@mail.gmail.com>

Could be an issue of export vs. set...
-MC

On Fri, Aug 3, 2012 at 1:03 PM, A E G <all.eforums at gmail.com> wrote:

> On Fri, Aug 3, 2012 at 3:51 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Did you solve it with setting a channel variable or a Sofia profile
>> setting? If a channel variable then you should be able to do a little
>> dialplan logic and set the value only if certain conditions are met, i.e.
>> if a certain gateway is going to be used.
>>
>> -MC
>>
>>
> Solved it using the Global variable in vars.xml. Sticking
> sdp_m_per_ptime=false as a channel variable (just before calling bridge)
> didn't seem to have any affect on the behaviour. Didn't try it in the
> specific profile to which that gateway belongs.
>
>
>> On Fri, Aug 3, 2012 at 12:44 PM, A E G <all.eforums at gmail.com> wrote:
>>
>>> ...and I'm not done yet.
>>>
>>> So while this one below solved the problem, how do we manage this so I
>>> don't always suppress all the m= lines in my SDP when sending calls to all
>>> the external gateways, but do that only when sending calls to any system
>>> that doesn't like it?
>>>
>>> As I said before, I tried doing this just before the bridge, but didn't
>>> work. Only seems to work as a PRE-PROCESS Global setting.
>>>
>>> Thx
>>>
>>> On Fri, Aug 3, 2012 at 1:44 AM, A E G <all.eforums at gmail.com> wrote:
>>>
>>>> Well turns out learning to use Google better always helps.
>>>>
>>>> Found a nugget of wisdom that <action application="set"
>>>> data="sdp_m_per_ptime=false"/> should've actually been in the vars.xml
>>>>
>>>> Seems to have fixed it.
>>>>
>>>>
>>>>
>>>> On Fri, Aug 3, 2012 at 12:01 AM, A E G <all.eforums at gmail.com> wrote:
>>>>
>>>>> Gents,
>>>>>
>>>>> running version: 1.2.0-rc2+git~20120731T213556Z~e97da8e20a
>>>>>
>>>>> have been fighting with this for an hour or more...have done a bit of
>>>>> research on the list and Google itself, scoured the Wiki etc. but can't
>>>>> seem to figure out where to set the codecs to be "well recd" by the remote
>>>>> SIP peer. This used to work fine until I upgraded 2 days ago and it stopped
>>>>> working without my doing / changing anything.
>>>>>
>>>>> The remote SIP peer is {*} which doesn't like multiple m= lines with
>>>>> differing ptime values.
>>>>>
>>>>> Debug on near-end says: "Originate Resulted in Error Cause: 88
>>>>> [INCOMPATIBLE_DESTINATION]"
>>>>>
>>>>> Full SDP here:
>>>>>
>>>>> v=0
>>>>> o=FreeSWITCH 1343934933 1343934934 IN IP4 192.168.1.80
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 192.168.1.80
>>>>> t=0 0
>>>>> m=audio 28884 RTP/AVP 98 8 3 101 13
>>>>> a=rtpmap:98 L16/16000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=ptime:20
>>>>> a=sendrecv
>>>>> m=audio 28884 RTP/AVP 0 101 13
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=ptime:40
>>>>> a=sendrecv
>>>>>
>>>>> Far end says:  Rejecting non-primary audio stream: audio 28884
>>>>> RTP/AVP 0 101 13
>>>>>
>>>>> I have tried to play around with the codec globals in vars.xml, to no
>>>>> avail.
>>>>>
>>>>> have also added stuff directly in the dialplan like so:
>>>>>
>>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>>> <action application="set" data="absolute_codec_string=PCMU at 40i"/>
>>>>> <action application="set" data="sdp_m_per_ptime=false"/>
>>>>> <action application="bridge" data="sofia/gateway/mycore1/1111"/>
>>>>>
>>>>>
>>>>> but no dice.
>>>>>
>>>>> Tried to remove the "absolute_codec_string", still no dice.
>>>>>
>>>>> I read about the "sdp_m_per_ptime"  that lets me "cheat" but that
>>>>> ain't doing anything either.
>>>>>
>>>>> What gives?
>>>>>
>>>>> Thx in advance
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From all.eforums at gmail.com  Sat Aug  4 00:49:13 2012
From: all.eforums at gmail.com (A E G)
Date: Fri, 3 Aug 2012 16:49:13 -0400
Subject: [Freeswitch-users] SOLVED: Asterisk don't like multiple m=
 lines in the SDP [Was: Error: 88 Incompatible Destination after upgrade]
In-Reply-To: <CAKzWOxU6PYAjafbef++inPFSU5Oc8VcK9=KE5rEwPQPhSti2Zg@mail.gmail.com>
References: <CADzY+JiUnUY+U=jrnVzBWNgYqp42jnEksHGA-DWfMnbt-Use9A@mail.gmail.com>
	<CADzY+JiUj=ExwEVL5hS0H=0uaNB9LjVoXXju0-mGejiTukK0ww@mail.gmail.com>
	<CAKzWOxVBbrTj1nWbqiT7=5d+LFUj6mkBTbKhieM7M+YHbW3CZw@mail.gmail.com>
	<CADzY+JiUDcq35rX2Y55JL_VwOnZ9ckKvrVNa50AudE7qOwqkWQ@mail.gmail.com>
	<CAKzWOxU6PYAjafbef++inPFSU5Oc8VcK9=KE5rEwPQPhSti2Zg@mail.gmail.com>
Message-ID: <CADzY+JiPhhPBnAa0ZRnEoqAb0Cn0p6uR3pkREWaGGuSHHEuQeg@mail.gmail.com>

On Fri, Aug 3, 2012 at 4:26 PM, Michael Collins <msc at freeswitch.org> wrote:

> Could be an issue of export vs. set...
> -MC
>
>
The Wiki says that export allows to carry over the value of a variable from
Leg A to Leg B. I am not sure this will solve my issue as I don't want to
export the codec offerings, more specifically varying ptime from Leg A to
Leg B. For this particular gateway, if I can, I only want to send
PCMU at 40i.Sorry if this is a stupid question, am an FS novice :)


>
> On Fri, Aug 3, 2012 at 1:03 PM, A E G <all.eforums at gmail.com> wrote:
>
>> On Fri, Aug 3, 2012 at 3:51 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Did you solve it with setting a channel variable or a Sofia profile
>>> setting? If a channel variable then you should be able to do a little
>>> dialplan logic and set the value only if certain conditions are met, i.e.
>>> if a certain gateway is going to be used.
>>>
>>> -MC
>>>
>>>
>> Solved it using the Global variable in vars.xml. Sticking
>> sdp_m_per_ptime=false as a channel variable (just before calling bridge)
>> didn't seem to have any affect on the behaviour. Didn't try it in the
>> specific profile to which that gateway belongs.
>>
>>
>>> On Fri, Aug 3, 2012 at 12:44 PM, A E G <all.eforums at gmail.com> wrote:
>>>
>>>> ...and I'm not done yet.
>>>>
>>>> So while this one below solved the problem, how do we manage this so I
>>>> don't always suppress all the m= lines in my SDP when sending calls to all
>>>> the external gateways, but do that only when sending calls to any system
>>>> that doesn't like it?
>>>>
>>>> As I said before, I tried doing this just before the bridge, but didn't
>>>> work. Only seems to work as a PRE-PROCESS Global setting.
>>>>
>>>> Thx
>>>>
>>>> On Fri, Aug 3, 2012 at 1:44 AM, A E G <all.eforums at gmail.com> wrote:
>>>>
>>>>> Well turns out learning to use Google better always helps.
>>>>>
>>>>> Found a nugget of wisdom that <action application="set"
>>>>> data="sdp_m_per_ptime=false"/> should've actually been in the vars.xml
>>>>>
>>>>> Seems to have fixed it.
>>>>>
>>>>>
>>>>>
>>>>> On Fri, Aug 3, 2012 at 12:01 AM, A E G <all.eforums at gmail.com> wrote:
>>>>>
>>>>>> Gents,
>>>>>>
>>>>>> running version: 1.2.0-rc2+git~20120731T213556Z~e97da8e20a
>>>>>>
>>>>>> have been fighting with this for an hour or more...have done a bit of
>>>>>> research on the list and Google itself, scoured the Wiki etc. but can't
>>>>>> seem to figure out where to set the codecs to be "well recd" by the remote
>>>>>> SIP peer. This used to work fine until I upgraded 2 days ago and it stopped
>>>>>> working without my doing / changing anything.
>>>>>>
>>>>>> The remote SIP peer is {*} which doesn't like multiple m= lines with
>>>>>> differing ptime values.
>>>>>>
>>>>>> Debug on near-end says: "Originate Resulted in Error Cause: 88
>>>>>> [INCOMPATIBLE_DESTINATION]"
>>>>>>
>>>>>> Full SDP here:
>>>>>>
>>>>>> v=0
>>>>>> o=FreeSWITCH 1343934933 1343934934 IN IP4 192.168.1.80
>>>>>> s=FreeSWITCH
>>>>>> c=IN IP4 192.168.1.80
>>>>>> t=0 0
>>>>>> m=audio 28884 RTP/AVP 98 8 3 101 13
>>>>>> a=rtpmap:98 L16/16000
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>> a=fmtp:101 0-16
>>>>>> a=ptime:20
>>>>>> a=sendrecv
>>>>>> m=audio 28884 RTP/AVP 0 101 13
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>> a=fmtp:101 0-16
>>>>>> a=ptime:40
>>>>>> a=sendrecv
>>>>>>
>>>>>> Far end says:  Rejecting non-primary audio stream: audio 28884
>>>>>> RTP/AVP 0 101 13
>>>>>>
>>>>>> I have tried to play around with the codec globals in vars.xml, to no
>>>>>> avail.
>>>>>>
>>>>>> have also added stuff directly in the dialplan like so:
>>>>>>
>>>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>>>> <action application="set" data="absolute_codec_string=PCMU at 40i"/>
>>>>>> <action application="set" data="sdp_m_per_ptime=false"/>
>>>>>> <action application="bridge" data="sofia/gateway/mycore1/1111"/>
>>>>>>
>>>>>>
>>>>>> but no dice.
>>>>>>
>>>>>> Tried to remove the "absolute_codec_string", still no dice.
>>>>>>
>>>>>> I read about the "sdp_m_per_ptime"  that lets me "cheat" but that
>>>>>> ain't doing anything either.
>>>>>>
>>>>>> What gives?
>>>>>>
>>>>>> Thx in advance
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Michael S Collins
>>> Twitter: @mercutioviz
>>> http://www.FreeSWITCH.org
>>> http://www.ClueCon.com
>>> http://www.OSTAG.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jackal at cybershroud.net  Sat Aug  4 00:52:39 2012
From: jackal at cybershroud.net (Carlos Flor)
Date: Fri, 3 Aug 2012 16:52:39 -0400
Subject: [Freeswitch-users] SIMPLE msg between two freeswitch servers
Message-ID: <CAAYhoXKvq-vXyi=RHWfUh6UDd_eY+9DcWFo4m9TVQa9u8V=Wuw@mail.gmail.com>

I am trying to send a SIP SIMPLE message from handset 101 registered to
switch A to handset 202 registered on switch C.  Switch B is in the middle
and is a gateway for switch A, so it looks something like this:

101 -----  FS A -----gw------- FS B -----------gw------- FS C --------- 202

I was able to get something working with the chatplan by setting the "to"
parameter to point to FS B and then on FS B setting the to parameter to
point to FS C.  I don't know if this is even the best way to do it, but
that's how I got it to work.  What I would like, however, is to use TLS on
all links.  I currently have TLS on all links for phone calls, but I guess
the chatplan doesn't use the same logic as the dialplan so by default, it
is using UDP between the switches.  I have been looking through the source
trying to find out what I can set to enable TLS for SIMPLE between switches
but I have been unsuccessful thus far.  Any help would be greatly
appreciated.

Thanks!
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From msc at freeswitch.org  Sat Aug  4 00:56:46 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 3 Aug 2012 13:56:46 -0700
Subject: [Freeswitch-users] SOLVED: Asterisk don't like multiple m=
 lines in the SDP [Was: Error: 88 Incompatible Destination after upgrade]
In-Reply-To: <CADzY+JiPhhPBnAa0ZRnEoqAb0Cn0p6uR3pkREWaGGuSHHEuQeg@mail.gmail.com>
References: <CADzY+JiUnUY+U=jrnVzBWNgYqp42jnEksHGA-DWfMnbt-Use9A@mail.gmail.com>
	<CADzY+JiUj=ExwEVL5hS0H=0uaNB9LjVoXXju0-mGejiTukK0ww@mail.gmail.com>
	<CAKzWOxVBbrTj1nWbqiT7=5d+LFUj6mkBTbKhieM7M+YHbW3CZw@mail.gmail.com>
	<CADzY+JiUDcq35rX2Y55JL_VwOnZ9ckKvrVNa50AudE7qOwqkWQ@mail.gmail.com>
	<CAKzWOxU6PYAjafbef++inPFSU5Oc8VcK9=KE5rEwPQPhSti2Zg@mail.gmail.com>
	<CADzY+JiPhhPBnAa0ZRnEoqAb0Cn0p6uR3pkREWaGGuSHHEuQeg@mail.gmail.com>
Message-ID: <CAKzWOxVYwsoniGQWGhSdWNxO5Hkr+fA87=iJKX=NzygC7kbHpQ@mail.gmail.com>

I'm talking about a specific channel variable that you say works when it's
set global but does not work when you only use the "set" app in the
dialplan. I recommend you take it out of vars.xml and then try using the
export application or put it in the dialstring using the
{chan_var=value}sofia/foo/bar at baz notation...

-MC

On Fri, Aug 3, 2012 at 1:49 PM, A E G <all.eforums at gmail.com> wrote:

> On Fri, Aug 3, 2012 at 4:26 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Could be an issue of export vs. set...
>> -MC
>>
>>
> The Wiki says that export allows to carry over the value of a variable
> from Leg A to Leg B. I am not sure this will solve my issue as I don't want
> to export the codec offerings, more specifically varying ptime from Leg A
> to Leg B. For this particular gateway, if I can, I only want to send
> PCMU at 40i. Sorry if this is a stupid question, am an FS novice :)
>
>
>>
>> On Fri, Aug 3, 2012 at 1:03 PM, A E G <all.eforums at gmail.com> wrote:
>>
>>> On Fri, Aug 3, 2012 at 3:51 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Did you solve it with setting a channel variable or a Sofia profile
>>>> setting? If a channel variable then you should be able to do a little
>>>> dialplan logic and set the value only if certain conditions are met, i.e.
>>>> if a certain gateway is going to be used.
>>>>
>>>> -MC
>>>>
>>>>
>>> Solved it using the Global variable in vars.xml. Sticking
>>> sdp_m_per_ptime=false as a channel variable (just before calling
>>> bridge) didn't seem to have any affect on the behaviour. Didn't try it in
>>> the specific profile to which that gateway belongs.
>>>
>>>
>>>> On Fri, Aug 3, 2012 at 12:44 PM, A E G <all.eforums at gmail.com> wrote:
>>>>
>>>>> ...and I'm not done yet.
>>>>>
>>>>> So while this one below solved the problem, how do we manage this so I
>>>>> don't always suppress all the m= lines in my SDP when sending calls to all
>>>>> the external gateways, but do that only when sending calls to any system
>>>>> that doesn't like it?
>>>>>
>>>>> As I said before, I tried doing this just before the bridge, but
>>>>> didn't work. Only seems to work as a PRE-PROCESS Global setting.
>>>>>
>>>>> Thx
>>>>>
>>>>> On Fri, Aug 3, 2012 at 1:44 AM, A E G <all.eforums at gmail.com> wrote:
>>>>>
>>>>>> Well turns out learning to use Google better always helps.
>>>>>>
>>>>>> Found a nugget of wisdom that <action application="set"
>>>>>> data="sdp_m_per_ptime=false"/> should've actually been in the
>>>>>> vars.xml
>>>>>>
>>>>>> Seems to have fixed it.
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Fri, Aug 3, 2012 at 12:01 AM, A E G <all.eforums at gmail.com> wrote:
>>>>>>
>>>>>>> Gents,
>>>>>>>
>>>>>>> running version: 1.2.0-rc2+git~20120731T213556Z~e97da8e20a
>>>>>>>
>>>>>>> have been fighting with this for an hour or more...have done a bit
>>>>>>> of research on the list and Google itself, scoured the Wiki etc. but can't
>>>>>>> seem to figure out where to set the codecs to be "well recd" by the remote
>>>>>>> SIP peer. This used to work fine until I upgraded 2 days ago and it stopped
>>>>>>> working without my doing / changing anything.
>>>>>>>
>>>>>>> The remote SIP peer is {*} which doesn't like multiple m= lines with
>>>>>>> differing ptime values.
>>>>>>>
>>>>>>> Debug on near-end says: "Originate Resulted in Error Cause: 88
>>>>>>> [INCOMPATIBLE_DESTINATION]"
>>>>>>>
>>>>>>> Full SDP here:
>>>>>>>
>>>>>>> v=0
>>>>>>> o=FreeSWITCH 1343934933 1343934934 IN IP4 192.168.1.80
>>>>>>> s=FreeSWITCH
>>>>>>> c=IN IP4 192.168.1.80
>>>>>>> t=0 0
>>>>>>> m=audio 28884 RTP/AVP 98 8 3 101 13
>>>>>>> a=rtpmap:98 L16/16000
>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>> a=fmtp:101 0-16
>>>>>>> a=ptime:20
>>>>>>> a=sendrecv
>>>>>>> m=audio 28884 RTP/AVP 0 101 13
>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>> a=fmtp:101 0-16
>>>>>>> a=ptime:40
>>>>>>> a=sendrecv
>>>>>>>
>>>>>>> Far end says:  Rejecting non-primary audio stream: audio 28884
>>>>>>> RTP/AVP 0 101 13
>>>>>>>
>>>>>>> I have tried to play around with the codec globals in vars.xml, to
>>>>>>> no avail.
>>>>>>>
>>>>>>> have also added stuff directly in the dialplan like so:
>>>>>>>
>>>>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>>>>> <action application="set" data="absolute_codec_string=PCMU at 40i"/>
>>>>>>> <action application="set" data="sdp_m_per_ptime=false"/>
>>>>>>> <action application="bridge" data="sofia/gateway/mycore1/1111"/>
>>>>>>>
>>>>>>>
>>>>>>> but no dice.
>>>>>>>
>>>>>>> Tried to remove the "absolute_codec_string", still no dice.
>>>>>>>
>>>>>>> I read about the "sdp_m_per_ptime"  that lets me "cheat" but that
>>>>>>> ain't doing anything either.
>>>>>>>
>>>>>>> What gives?
>>>>>>>
>>>>>>> Thx in advance
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Michael S Collins
>>>> Twitter: @mercutioviz
>>>> http://www.FreeSWITCH.org
>>>> http://www.ClueCon.com
>>>> http://www.OSTAG.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From all.eforums at gmail.com  Sat Aug  4 01:08:27 2012
From: all.eforums at gmail.com (A E G)
Date: Fri, 3 Aug 2012 17:08:27 -0400
Subject: [Freeswitch-users] SOLVED: Asterisk don't like multiple m=
 lines in the SDP [Was: Error: 88 Incompatible Destination after upgrade]
In-Reply-To: <CAKzWOxVYwsoniGQWGhSdWNxO5Hkr+fA87=iJKX=NzygC7kbHpQ@mail.gmail.com>
References: <CADzY+JiUnUY+U=jrnVzBWNgYqp42jnEksHGA-DWfMnbt-Use9A@mail.gmail.com>
	<CADzY+JiUj=ExwEVL5hS0H=0uaNB9LjVoXXju0-mGejiTukK0ww@mail.gmail.com>
	<CAKzWOxVBbrTj1nWbqiT7=5d+LFUj6mkBTbKhieM7M+YHbW3CZw@mail.gmail.com>
	<CADzY+JiUDcq35rX2Y55JL_VwOnZ9ckKvrVNa50AudE7qOwqkWQ@mail.gmail.com>
	<CAKzWOxU6PYAjafbef++inPFSU5Oc8VcK9=KE5rEwPQPhSti2Zg@mail.gmail.com>
	<CADzY+JiPhhPBnAa0ZRnEoqAb0Cn0p6uR3pkREWaGGuSHHEuQeg@mail.gmail.com>
	<CAKzWOxVYwsoniGQWGhSdWNxO5Hkr+fA87=iJKX=NzygC7kbHpQ@mail.gmail.com>
Message-ID: <CADzY+JhqXK7d08BUwHdiKFLVDXxxFS30gjayfhzxgddLL6n0HA@mail.gmail.com>

On Fri, Aug 3, 2012 at 4:56 PM, Michael Collins <msc at freeswitch.org> wrote:

> I'm talking about a specific channel variable that you say works when it's
> set global but does not work when you only use the "set" app in the
> dialplan. I recommend you take it out of vars.xml and then try using the
> export application or put it in the dialstring using the
> {chan_var=value}sofia/foo/bar at baz notation...
>
> -MC
>
>
Ok, so that's what I was going to ask; is setting the channel variable in
the dialstring different than setting it within the dialplan just before
bridge is called. I guess Leg B doesn't really start until the Bridge is
called ...duh! Setting the {chan_var} in the dialstring worked. Thanks!


>
> On Fri, Aug 3, 2012 at 1:49 PM, A E G <all.eforums at gmail.com> wrote:
>
>> On Fri, Aug 3, 2012 at 4:26 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Could be an issue of export vs. set...
>>> -MC
>>>
>>>
>> The Wiki says that export allows to carry over the value of a variable
>> from Leg A to Leg B. I am not sure this will solve my issue as I don't want
>> to export the codec offerings, more specifically varying ptime from Leg A
>> to Leg B. For this particular gateway, if I can, I only want to send
>> PCMU at 40i. Sorry if this is a stupid question, am an FS novice :)
>>
>>
>>>
>>> On Fri, Aug 3, 2012 at 1:03 PM, A E G <all.eforums at gmail.com> wrote:
>>>
>>>> On Fri, Aug 3, 2012 at 3:51 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> Did you solve it with setting a channel variable or a Sofia profile
>>>>> setting? If a channel variable then you should be able to do a little
>>>>> dialplan logic and set the value only if certain conditions are met, i.e.
>>>>> if a certain gateway is going to be used.
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>> Solved it using the Global variable in vars.xml. Sticking
>>>> sdp_m_per_ptime=false as a channel variable (just before calling
>>>> bridge) didn't seem to have any affect on the behaviour. Didn't try it in
>>>> the specific profile to which that gateway belongs.
>>>>
>>>>
>>>>> On Fri, Aug 3, 2012 at 12:44 PM, A E G <all.eforums at gmail.com> wrote:
>>>>>
>>>>>> ...and I'm not done yet.
>>>>>>
>>>>>> So while this one below solved the problem, how do we manage this so
>>>>>> I don't always suppress all the m= lines in my SDP when sending calls to
>>>>>> all the external gateways, but do that only when sending calls to any
>>>>>> system that doesn't like it?
>>>>>>
>>>>>> As I said before, I tried doing this just before the bridge, but
>>>>>> didn't work. Only seems to work as a PRE-PROCESS Global setting.
>>>>>>
>>>>>> Thx
>>>>>>
>>>>>> On Fri, Aug 3, 2012 at 1:44 AM, A E G <all.eforums at gmail.com> wrote:
>>>>>>
>>>>>>> Well turns out learning to use Google better always helps.
>>>>>>>
>>>>>>> Found a nugget of wisdom that <action application="set"
>>>>>>> data="sdp_m_per_ptime=false"/> should've actually been in the
>>>>>>> vars.xml
>>>>>>>
>>>>>>> Seems to have fixed it.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Fri, Aug 3, 2012 at 12:01 AM, A E G <all.eforums at gmail.com>wrote:
>>>>>>>
>>>>>>>> Gents,
>>>>>>>>
>>>>>>>> running version: 1.2.0-rc2+git~20120731T213556Z~e97da8e20a
>>>>>>>>
>>>>>>>> have been fighting with this for an hour or more...have done a bit
>>>>>>>> of research on the list and Google itself, scoured the Wiki etc. but can't
>>>>>>>> seem to figure out where to set the codecs to be "well recd" by the remote
>>>>>>>> SIP peer. This used to work fine until I upgraded 2 days ago and it stopped
>>>>>>>> working without my doing / changing anything.
>>>>>>>>
>>>>>>>> The remote SIP peer is {*} which doesn't like multiple m= lines
>>>>>>>> with differing ptime values.
>>>>>>>>
>>>>>>>> Debug on near-end says: "Originate Resulted in Error Cause: 88
>>>>>>>> [INCOMPATIBLE_DESTINATION]"
>>>>>>>>
>>>>>>>> Full SDP here:
>>>>>>>>
>>>>>>>> v=0
>>>>>>>> o=FreeSWITCH 1343934933 1343934934 IN IP4 192.168.1.80
>>>>>>>> s=FreeSWITCH
>>>>>>>> c=IN IP4 192.168.1.80
>>>>>>>> t=0 0
>>>>>>>> m=audio 28884 RTP/AVP 98 8 3 101 13
>>>>>>>> a=rtpmap:98 L16/16000
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>> a=fmtp:101 0-16
>>>>>>>> a=ptime:20
>>>>>>>> a=sendrecv
>>>>>>>> m=audio 28884 RTP/AVP 0 101 13
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>> a=fmtp:101 0-16
>>>>>>>> a=ptime:40
>>>>>>>> a=sendrecv
>>>>>>>>
>>>>>>>> Far end says:  Rejecting non-primary audio stream: audio 28884
>>>>>>>> RTP/AVP 0 101 13
>>>>>>>>
>>>>>>>> I have tried to play around with the codec globals in vars.xml, to
>>>>>>>> no avail.
>>>>>>>>
>>>>>>>> have also added stuff directly in the dialplan like so:
>>>>>>>>
>>>>>>>> <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>> <action application="set" data="absolute_codec_string=PCMU at 40i"/>
>>>>>>>> <action application="set" data="sdp_m_per_ptime=false"/>
>>>>>>>> <action application="bridge" data="sofia/gateway/mycore1/1111"/>
>>>>>>>>
>>>>>>>>
>>>>>>>> but no dice.
>>>>>>>>
>>>>>>>> Tried to remove the "absolute_codec_string", still no dice.
>>>>>>>>
>>>>>>>> I read about the "sdp_m_per_ptime"  that lets me "cheat" but that
>>>>>>>> ain't doing anything either.
>>>>>>>>
>>>>>>>> What gives?
>>>>>>>>
>>>>>>>> Thx in advance
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Michael S Collins
>>>>> Twitter: @mercutioviz
>>>>> http://www.FreeSWITCH.org
>>>>> http://www.ClueCon.com
>>>>> http://www.OSTAG.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Michael S Collins
>>> Twitter: @mercutioviz
>>> http://www.FreeSWITCH.org
>>> http://www.ClueCon.com
>>> http://www.OSTAG.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ben at langfeld.co.uk  Sat Aug  4 01:57:30 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 3 Aug 2012 22:57:30 +0100
Subject: [Freeswitch-users] Speak application w/ SSML over IES
Message-ID: <CAAyX+KF0cVq=LZ2ixSxFkOz+WOBwjt3mZzgRzcMMTLi2VKUe-Q@mail.gmail.com>

Attempting the following via IES results in flite reading out the XML
document, rather than rendering it as SSML, regardless of what
escaping I try. Can anyone clarify how this is supposed to be done?

SendMsg 0e66ee48-ddb6-11e1-a46e-b7cac2dab4cc
call-command: execute
execute-app-name: speak
execute-app-arg:
%[punchblock_component_id=e8f7ebcb-d107-4d6e-bf9f-fca4d7b50d01]flite|kal|<speak
xmlns="http://www.w3.org/2001/10/synthesis" version="1.0"
xml:lang="en-US">Hello Fernando</speak>

Regards,
Ben Langfeld


From cmrienzo at gmail.com  Sat Aug  4 02:36:29 2012
From: cmrienzo at gmail.com (cmrienzo at gmail.com)
Date: Fri, 3 Aug 2012 18:36:29 -0400
Subject: [Freeswitch-users] Speak application w/ SSML over IES
In-Reply-To: <CAAyX+KF0cVq=LZ2ixSxFkOz+WOBwjt3mZzgRzcMMTLi2VKUe-Q@mail.gmail.com>
References: <CAAyX+KF0cVq=LZ2ixSxFkOz+WOBwjt3mZzgRzcMMTLi2VKUe-Q@mail.gmail.com>
Message-ID: <F707213C-8CC1-4DAB-ACDE-2BA4FDDDA2C6@gmail.com>

I'm pretty sure flite only supports plain text. 


On Aug 3, 2012, at 17:57, Ben Langfeld <ben at langfeld.co.uk> wrote:

> Attempting the following via IES results in flite reading out the XML
> document, rather than rendering it as SSML, regardless of what
> escaping I try. Can anyone clarify how this is supposed to be done?
> 
> SendMsg 0e66ee48-ddb6-11e1-a46e-b7cac2dab4cc
> call-command: execute
> execute-app-name: speak
> execute-app-arg:
> %[punchblock_component_id=e8f7ebcb-d107-4d6e-bf9f-fca4d7b50d01]flite|kal|<speak
> xmlns="http://www.w3.org/2001/10/synthesis" version="1.0"
> xml:lang="en-US">Hello Fernando</speak>
> 
> Regards,
> Ben Langfeld
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From gabe at gundy.org  Sat Aug  4 11:11:29 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 4 Aug 2012 01:11:29 -0600
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 73, Issue 228
In-Reply-To: <CANNhuhd_nTpr1vmwT6_FeVQY2Z7bs7Jq8RpOPPL31j2myfa36g@mail.gmail.com>
References: <mailman.15633.1343724566.13443.freeswitch-users@lists.freeswitch.org>
	<CANNhuhd_nTpr1vmwT6_FeVQY2Z7bs7Jq8RpOPPL31j2myfa36g@mail.gmail.com>
Message-ID: <CAMwZ-tL=jA+U+8i7jtUs0caOVkt9uikjGhWg6KvoJGvcwWtccA@mail.gmail.com>

On Tue, Jul 31, 2012 at 4:58 AM, virendra bhati <virbhati at gmail.com> wrote:
> Yes
>
> ignore_early_media=true
>
> no improvement

I, for one, have no idea what you're talking about. If you're looking
for help, please make it easy on us.

* Don't reply to email digests.
* Do trim the email.
* Consider bottom posting *if* it adds clarity  (please no top vs.
bottom posting follow ups).
* Reply in full sentences.


All of these things will show you respect the efforts of the many
users who freely give their time to help out on this list.

Good luck going forward.


Best,
Gabe


From sdevoy at bizfocused.com  Sat Aug  4 21:22:36 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Sat, 4 Aug 2012 13:22:36 -0400
Subject: [Freeswitch-users] Odd console messages
Message-ID: <165e01cd7265$be2e6f70$3a8b4e50$@bizfocused.com>

HI All,

 

Saw something new today and I am kind of worried that something has been
hacked.

 

What does this mean in the console, it is during an incoming fax to rxfax:

DICK 6890

DICK 6896

WTFX
Replaces=49ed5a45-dcb3bb11%40192.168.0.24%3Bfrom-tag%3D3e5df0b5191e9761o3%3B
to-tag%3Djac81BFj6rF7e

DICK 6890

DICK 6896

WTFX
Replaces=7ad19169-23089a56%40192.168.0.16%3Bfrom-tag%3D82d8de4d189d12c2o3%3B
to-tag%3DN15FBmN0yyt6Q

DICK 6890

DICK 6896

WTFX
Replaces=200b9844-d8cf82cc%40192.168.0.24%3Bfrom-tag%3Daf559e4ebc7adaco3%3Bt
o-tag%3DUtBZUyQF90etQ

DICK 6890

DICK 6896

WTFX
Replaces=f89a0068-6396bf3c%40192.168.0.16%3Bfrom-tag%3Df9aaee7897c4286co3%3B
to-tag%3D6Bg2yD8j8jjeH

DICK 6890

DICK 6896

WTFX
Replaces=3d6e5fdd-a0216b6d%40192.168.0.16%3Bfrom-tag%3D5517551d4feaf72do3%3B
to-tag%3DXB17S1pHp3X8a

DICK 6890

DICK 6896

WTFX
Replaces=f7ddd747-59b1035b%40192.168.0.16%3Bfrom-tag%3D717fa657bf1aff8bo3%3B
to-tag%3DjN1c6c27027Sr

DICK 6890

DICK 6896

WTFX
Replaces=e5fff425-65d08882%40192.168.0.24%3Bfrom-tag%3D716f37c97bdf792eo3%3B
to-tag%3DjcjZXKNDF7ZQg

DICK 6890

DICK 6896

WTFX
Replaces=798498b-60fe2ae9%40192.168.0.24%3Bfrom-tag%3Df86b8923d68e3031o3%3Bt
o-tag%3DHcyymrUy2a0Za

DICK 6890

DICK 6896

WTFX
Replaces=49ed5a45-dcb3bb11%40192.168.0.24%3Bfrom-tag%3D3e5df0b5191e9761o3%3B
to-tag%3Djac81BFj6rF7e

DICK 6890

DICK 6896

WTFX
Replaces=7ad19169-23089a56%40192.168.0.16%3Bfrom-tag%3D82d8de4d189d12c2o3%3B
to-tag%3DN15FBmN0yyt6Q

DICK 6890

DICK 6896

WTFX
Replaces=200b9844-d8cf82cc%40192.168.0.24%3Bfrom-tag%3Daf559e4ebc7adaco3%3Bt
o-tag%3DUtBZUyQF90etQ

DICK 6890

DICK 6896

WTFX
Replaces=f89a0068-6396bf3c%40192.168.0.16%3Bfrom-tag%3Df9aaee7897c4286co3%3B
to-tag%3D6Bg2yD8j8jjeH

DICK 6890

DICK 6896

WTFX
Replaces=3d6e5fdd-a0216b6d%40192.168.0.16%3Bfrom-tag%3D5517551d4feaf72do3%3B
to-tag%3DXB17S1pHp3X8a

DICK 6890

DICK 6896

WTFX
Replaces=f7ddd747-59b1035b%40192.168.0.16%3Bfrom-tag%3D717fa657bf1aff8bo3%3B
to-tag%3DjN1c6c27027Sr

DICK 6890

DICK 6896

WTFX
Replaces=e5fff425-65d08882%40192.168.0.24%3Bfrom-tag%3D716f37c97bdf792eo3%3B
to-tag%3DjcjZXKNDF7ZQg

DICK 6890

DICK 6896

WTFX
Replaces=798498b-60fe2ae9%40192.168.0.24%3Bfrom-tag%3Df86b8923d68e3031o3%3Bt
o-tag%3DHcyymrUy2a0Za

 

2012-08-04 11:13:46.491588 [DEBUG] switch_core_state_machine.c:416
(sofia/external_noauth/.

 

Anyone seen those before?

 

Sean

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From william.king at quentustech.com  Sat Aug  4 21:29:19 2012
From: william.king at quentustech.com (William King)
Date: Sat, 04 Aug 2012 13:29:19 -0400
Subject: [Freeswitch-users] Odd console messages
In-Reply-To: <165e01cd7265$be2e6f70$3a8b4e50$@bizfocused.com>
References: <165e01cd7265$be2e6f70$3a8b4e50$@bizfocused.com>
Message-ID: <501D5BEF.5040301@quentustech.com>

Run make current and update your code.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 08/04/2012 01:22 PM, Sean Devoy wrote:
>
> HI All,
>
> Saw something new today and I am kind of worried that something has 
> been hacked.
>
> What does this mean in the console, it is during an incoming fax to rxfax:
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=49ed5a45-dcb3bb11%40192.168.0.24%3Bfrom-tag%3D3e5df0b5191e9761o3%3Bto-tag%3Djac81BFj6rF7e
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=7ad19169-23089a56%40192.168.0.16%3Bfrom-tag%3D82d8de4d189d12c2o3%3Bto-tag%3DN15FBmN0yyt6Q
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=200b9844-d8cf82cc%40192.168.0.24%3Bfrom-tag%3Daf559e4ebc7adaco3%3Bto-tag%3DUtBZUyQF90etQ
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=f89a0068-6396bf3c%40192.168.0.16%3Bfrom-tag%3Df9aaee7897c4286co3%3Bto-tag%3D6Bg2yD8j8jjeH
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=3d6e5fdd-a0216b6d%40192.168.0.16%3Bfrom-tag%3D5517551d4feaf72do3%3Bto-tag%3DXB17S1pHp3X8a
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=f7ddd747-59b1035b%40192.168.0.16%3Bfrom-tag%3D717fa657bf1aff8bo3%3Bto-tag%3DjN1c6c27027Sr
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=e5fff425-65d08882%40192.168.0.24%3Bfrom-tag%3D716f37c97bdf792eo3%3Bto-tag%3DjcjZXKNDF7ZQg
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=798498b-60fe2ae9%40192.168.0.24%3Bfrom-tag%3Df86b8923d68e3031o3%3Bto-tag%3DHcyymrUy2a0Za
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=49ed5a45-dcb3bb11%40192.168.0.24%3Bfrom-tag%3D3e5df0b5191e9761o3%3Bto-tag%3Djac81BFj6rF7e
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=7ad19169-23089a56%40192.168.0.16%3Bfrom-tag%3D82d8de4d189d12c2o3%3Bto-tag%3DN15FBmN0yyt6Q
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=200b9844-d8cf82cc%40192.168.0.24%3Bfrom-tag%3Daf559e4ebc7adaco3%3Bto-tag%3DUtBZUyQF90etQ
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=f89a0068-6396bf3c%40192.168.0.16%3Bfrom-tag%3Df9aaee7897c4286co3%3Bto-tag%3D6Bg2yD8j8jjeH
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=3d6e5fdd-a0216b6d%40192.168.0.16%3Bfrom-tag%3D5517551d4feaf72do3%3Bto-tag%3DXB17S1pHp3X8a
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=f7ddd747-59b1035b%40192.168.0.16%3Bfrom-tag%3D717fa657bf1aff8bo3%3Bto-tag%3DjN1c6c27027Sr
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=e5fff425-65d08882%40192.168.0.24%3Bfrom-tag%3D716f37c97bdf792eo3%3Bto-tag%3DjcjZXKNDF7ZQg
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=798498b-60fe2ae9%40192.168.0.24%3Bfrom-tag%3Df86b8923d68e3031o3%3Bto-tag%3DHcyymrUy2a0Za
>
> 2012-08-04 11:13:46.491588 [DEBUG] switch_core_state_machine.c:416 
> (sofia/external_noauth/...
>
> Anyone seen those before?
>
> Sean
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.jugatsu at gmail.com  Sun Aug  5 00:05:08 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Sun, 5 Aug 2012 00:05:08 +0400
Subject: [Freeswitch-users] Odd console messages
In-Reply-To: <165e01cd7265$be2e6f70$3a8b4e50$@bizfocused.com>
References: <165e01cd7265$be2e6f70$3a8b4e50$@bizfocused.com>
Message-ID: <CANBhyPBu4RNg7vqX3TKPL4U6Ywp2e43wj7zbgG75U5o=AgxXzw@mail.gmail.com>

It's quite funny. Can you please post all debug log to pastebin.
04.08.2012 21:24 ???????????? "Sean Devoy" <sdevoy at bizfocused.com> ???????:

> HI All,****
>
> ** **
>
> Saw something new today and I am kind of worried that something has been
> hacked.****
>
> ** **
>
> What does this mean in the console, it is during an incoming fax to rxfax:
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=49ed5a45-dcb3bb11%40192.168.0.24%3Bfrom-tag%3D3e5df0b5191e9761o3%3Bto-tag%3Djac81BFj6rF7e
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=7ad19169-23089a56%40192.168.0.16%3Bfrom-tag%3D82d8de4d189d12c2o3%3Bto-tag%3DN15FBmN0yyt6Q
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=200b9844-d8cf82cc%40192.168.0.24%3Bfrom-tag%3Daf559e4ebc7adaco3%3Bto-tag%3DUtBZUyQF90etQ
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=f89a0068-6396bf3c%40192.168.0.16%3Bfrom-tag%3Df9aaee7897c4286co3%3Bto-tag%3D6Bg2yD8j8jjeH
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=3d6e5fdd-a0216b6d%40192.168.0.16%3Bfrom-tag%3D5517551d4feaf72do3%3Bto-tag%3DXB17S1pHp3X8a
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=f7ddd747-59b1035b%40192.168.0.16%3Bfrom-tag%3D717fa657bf1aff8bo3%3Bto-tag%3DjN1c6c27027Sr
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=e5fff425-65d08882%40192.168.0.24%3Bfrom-tag%3D716f37c97bdf792eo3%3Bto-tag%3DjcjZXKNDF7ZQg
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=798498b-60fe2ae9%40192.168.0.24%3Bfrom-tag%3Df86b8923d68e3031o3%3Bto-tag%3DHcyymrUy2a0Za
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=49ed5a45-dcb3bb11%40192.168.0.24%3Bfrom-tag%3D3e5df0b5191e9761o3%3Bto-tag%3Djac81BFj6rF7e
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=7ad19169-23089a56%40192.168.0.16%3Bfrom-tag%3D82d8de4d189d12c2o3%3Bto-tag%3DN15FBmN0yyt6Q
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=200b9844-d8cf82cc%40192.168.0.24%3Bfrom-tag%3Daf559e4ebc7adaco3%3Bto-tag%3DUtBZUyQF90etQ
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=f89a0068-6396bf3c%40192.168.0.16%3Bfrom-tag%3Df9aaee7897c4286co3%3Bto-tag%3D6Bg2yD8j8jjeH
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=3d6e5fdd-a0216b6d%40192.168.0.16%3Bfrom-tag%3D5517551d4feaf72do3%3Bto-tag%3DXB17S1pHp3X8a
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=f7ddd747-59b1035b%40192.168.0.16%3Bfrom-tag%3D717fa657bf1aff8bo3%3Bto-tag%3DjN1c6c27027Sr
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=e5fff425-65d08882%40192.168.0.24%3Bfrom-tag%3D716f37c97bdf792eo3%3Bto-tag%3DjcjZXKNDF7ZQg
> ****
>
> DICK 6890****
>
> DICK 6896****
>
> WTFX
> Replaces=798498b-60fe2ae9%40192.168.0.24%3Bfrom-tag%3Df86b8923d68e3031o3%3Bto-tag%3DHcyymrUy2a0Za
> ****
>
> ** **
>
> 2012-08-04 11:13:46.491588 [DEBUG] switch_core_state_machine.c:416
> (sofia/external_noauth/...****
>
> ** **
>
> Anyone seen those before?****
>
> ** **
>
> Sean****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bwibowo at gmail.com  Sun Aug  5 02:12:56 2012
From: bwibowo at gmail.com (budi wibowo)
Date: Sun, 5 Aug 2012 05:12:56 +0700
Subject: [Freeswitch-users] sip external port
Message-ID: <CAKnRR0YB92Q+w0WQQeTSX+WKiiAQD7Nr6vgiXSYz0UbBNJUcJQ@mail.gmail.com>

hi
just curious why freeswitch using port 5080 instead of 5060 for
communication to external sip server.
few days ago i interconnect fs to external sip server that running on port
5060, and this port id fixed.
how to overcome this situation? other pbx like asterisk can use port 5060
for both internal and external


regards

budi
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From curriegrad2004 at gmail.com  Sun Aug  5 03:07:12 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 4 Aug 2012 16:07:12 -0700
Subject: [Freeswitch-users] sip external port
In-Reply-To: <CAKnRR0YB92Q+w0WQQeTSX+WKiiAQD7Nr6vgiXSYz0UbBNJUcJQ@mail.gmail.com>
References: <CAKnRR0YB92Q+w0WQQeTSX+WKiiAQD7Nr6vgiXSYz0UbBNJUcJQ@mail.gmail.com>
Message-ID: <CAMWiMy8zSzOrvH8FcZmAybHgSoYhA_vLyxtFnPLyJmnE+M3SnQ@mail.gmail.com>

FreeSWITCH uses port 5080 for internal phones to register for security
reasons. Although 5060 is the default port I'd advise you to leave it as is
for obvious reasons.
On Aug 4, 2012 3:14 PM, "budi wibowo" <bwibowo at gmail.com> wrote:

> hi
> just curious why freeswitch using port 5080 instead of 5060 for
> communication to external sip server.
> few days ago i interconnect fs to external sip server that running on port
> 5060, and this port id fixed.
> how to overcome this situation? other pbx like asterisk can use port 5060
> for both internal and external
>
>
> regards
>
> budi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sdevoy at bizfocused.com  Sun Aug  5 05:35:29 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Sat, 4 Aug 2012 21:35:29 -0400
Subject: [Freeswitch-users] Odd console messages
In-Reply-To: <501D5BEF.5040301@quentustech.com>
References: <165e01cd7265$be2e6f70$3a8b4e50$@bizfocused.com>
	<501D5BEF.5040301@quentustech.com>
Message-ID: <17bf01cd72aa$98d62430$ca826c90$@bizfocused.com>

Sorry, that sounds a little too generic for me.  Besides this is a
production server and with the instability of the head in recent weeks I
cannot do that without SOME testing on another server.

 

Do you have a reason to make current, or is that just a catch all?

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William
King
Sent: Saturday, August 04, 2012 1:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Odd console messages

 

Run make current and update your code. 



 
William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com 


On 08/04/2012 01:22 PM, Sean Devoy wrote: 

HI All,

 

Saw something new today and I am kind of worried that something has been
hacked.

 

What does this mean in the console, it is during an incoming fax to rxfax:

DICK 6890

DICK 6896

WTFX
Replaces=49ed5a45-dcb3bb11%40192.168.0.24%3Bfrom-tag%3D3e5df0b5191e9761o3%3B
to-tag%3Djac81BFj6rF7e

DICK 6890

DICK 6896

WTFX
Replaces=7ad19169-23089a56%40192.168.0.16%3Bfrom-tag%3D82d8de4d189d12c2o3%3B
to-tag%3DN15FBmN0yyt6Q

DICK 6890

DICK 6896

WTFX
Replaces=200b9844-d8cf82cc%40192.168.0.24%3Bfrom-tag%3Daf559e4ebc7adaco3%3Bt
o-tag%3DUtBZUyQF90etQ

DICK 6890

DICK 6896

WTFX
Replaces=f89a0068-6396bf3c%40192.168.0.16%3Bfrom-tag%3Df9aaee7897c4286co3%3B
to-tag%3D6Bg2yD8j8jjeH

DICK 6890

DICK 6896

WTFX
Replaces=3d6e5fdd-a0216b6d%40192.168.0.16%3Bfrom-tag%3D5517551d4feaf72do3%3B
to-tag%3DXB17S1pHp3X8a

DICK 6890

DICK 6896

WTFX
Replaces=f7ddd747-59b1035b%40192.168.0.16%3Bfrom-tag%3D717fa657bf1aff8bo3%3B
to-tag%3DjN1c6c27027Sr

DICK 6890

DICK 6896

WTFX
Replaces=e5fff425-65d08882%40192.168.0.24%3Bfrom-tag%3D716f37c97bdf792eo3%3B
to-tag%3DjcjZXKNDF7ZQg

DICK 6890

DICK 6896

WTFX
Replaces=798498b-60fe2ae9%40192.168.0.24%3Bfrom-tag%3Df86b8923d68e3031o3%3Bt
o-tag%3DHcyymrUy2a0Za

DICK 6890

DICK 6896

WTFX
Replaces=49ed5a45-dcb3bb11%40192.168.0.24%3Bfrom-tag%3D3e5df0b5191e9761o3%3B
to-tag%3Djac81BFj6rF7e

DICK 6890

DICK 6896

WTFX
Replaces=7ad19169-23089a56%40192.168.0.16%3Bfrom-tag%3D82d8de4d189d12c2o3%3B
to-tag%3DN15FBmN0yyt6Q

DICK 6890

DICK 6896

WTFX
Replaces=200b9844-d8cf82cc%40192.168.0.24%3Bfrom-tag%3Daf559e4ebc7adaco3%3Bt
o-tag%3DUtBZUyQF90etQ

DICK 6890

DICK 6896

WTFX
Replaces=f89a0068-6396bf3c%40192.168.0.16%3Bfrom-tag%3Df9aaee7897c4286co3%3B
to-tag%3D6Bg2yD8j8jjeH

DICK 6890

DICK 6896

WTFX
Replaces=3d6e5fdd-a0216b6d%40192.168.0.16%3Bfrom-tag%3D5517551d4feaf72do3%3B
to-tag%3DXB17S1pHp3X8a

DICK 6890

DICK 6896

WTFX
Replaces=f7ddd747-59b1035b%40192.168.0.16%3Bfrom-tag%3D717fa657bf1aff8bo3%3B
to-tag%3DjN1c6c27027Sr

DICK 6890

DICK 6896

WTFX
Replaces=e5fff425-65d08882%40192.168.0.24%3Bfrom-tag%3D716f37c97bdf792eo3%3B
to-tag%3DjcjZXKNDF7ZQg

DICK 6890

DICK 6896

WTFX
Replaces=798498b-60fe2ae9%40192.168.0.24%3Bfrom-tag%3Df86b8923d68e3031o3%3Bt
o-tag%3DHcyymrUy2a0Za

 

2012-08-04 11:13:46.491588 [DEBUG] switch_core_state_machine.c:416
(sofia/external_noauth/.

 

Anyone seen those before?

 

Sean






_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com
 


 
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
 
Join Us At ClueCon - Aug 7-9, 2012
 
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From sdevoy at bizfocused.com  Sun Aug  5 05:43:04 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Sat, 4 Aug 2012 21:43:04 -0400
Subject: [Freeswitch-users] Odd console messages
In-Reply-To: <CANBhyPBu4RNg7vqX3TKPL4U6Ywp2e43wj7zbgG75U5o=AgxXzw@mail.gmail.com>
References: <165e01cd7265$be2e6f70$3a8b4e50$@bizfocused.com>
	<CANBhyPBu4RNg7vqX3TKPL4U6Ywp2e43wj7zbgG75U5o=AgxXzw@mail.gmail.com>
Message-ID: <17cd01cd72ab$a7ec55b0$f7c50110$@bizfocused.com>

Here is the complete log of that call.  If you want the whole 5MB say so.

 

http://pastebin.freeswitch.com/19639

 

I mean people call me that all the time, but Freeswitch never has!

 

Thanks,

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anton
Kvashenkin
Sent: Saturday, August 04, 2012 4:05 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Odd console messages

 

It's quite funny. Can you please post all debug log to pastebin.

04.08.2012 21:24 ???????????? "Sean Devoy" <sdevoy at bizfocused.com> ???????:

HI All,

 

Saw something new today and I am kind of worried that something has been
hacked.

 

What does this mean in the console, it is during an incoming fax to rxfax:

DICK 6890

DICK 6896

WTFX
Replaces=49ed5a45-dcb3bb11%40192.168.0.24%3Bfrom-tag%3D3e5df0b5191e9761o3%3B
to-tag%3Djac81BFj6rF7e

DICK 6890

DICK 6896

WTFX
Replaces=7ad19169-23089a56%40192.168.0.16%3Bfrom-tag%3D82d8de4d189d12c2o3%3B
to-tag%3DN15FBmN0yyt6Q

DICK 6890

DICK 6896

WTFX
Replaces=200b9844-d8cf82cc%40192.168.0.24%3Bfrom-tag%3Daf559e4ebc7adaco3%3Bt
o-tag%3DUtBZUyQF90etQ

DICK 6890

DICK 6896

WTFX
Replaces=f89a0068-6396bf3c%40192.168.0.16%3Bfrom-tag%3Df9aaee7897c4286co3%3B
to-tag%3D6Bg2yD8j8jjeH

DICK 6890

DICK 6896

WTFX
Replaces=3d6e5fdd-a0216b6d%40192.168.0.16%3Bfrom-tag%3D5517551d4feaf72do3%3B
to-tag%3DXB17S1pHp3X8a

DICK 6890

DICK 6896

WTFX
Replaces=f7ddd747-59b1035b%40192.168.0.16%3Bfrom-tag%3D717fa657bf1aff8bo3%3B
to-tag%3DjN1c6c27027Sr

DICK 6890

DICK 6896

WTFX
Replaces=e5fff425-65d08882%40192.168.0.24%3Bfrom-tag%3D716f37c97bdf792eo3%3B
to-tag%3DjcjZXKNDF7ZQg

DICK 6890

DICK 6896

WTFX
Replaces=798498b-60fe2ae9%40192.168.0.24%3Bfrom-tag%3Df86b8923d68e3031o3%3Bt
o-tag%3DHcyymrUy2a0Za

DICK 6890

DICK 6896

WTFX
Replaces=49ed5a45-dcb3bb11%40192.168.0.24%3Bfrom-tag%3D3e5df0b5191e9761o3%3B
to-tag%3Djac81BFj6rF7e

DICK 6890

DICK 6896

WTFX
Replaces=7ad19169-23089a56%40192.168.0.16%3Bfrom-tag%3D82d8de4d189d12c2o3%3B
to-tag%3DN15FBmN0yyt6Q

DICK 6890

DICK 6896

WTFX
Replaces=200b9844-d8cf82cc%40192.168.0.24%3Bfrom-tag%3Daf559e4ebc7adaco3%3Bt
o-tag%3DUtBZUyQF90etQ

DICK 6890

DICK 6896

WTFX
Replaces=f89a0068-6396bf3c%40192.168.0.16%3Bfrom-tag%3Df9aaee7897c4286co3%3B
to-tag%3D6Bg2yD8j8jjeH

DICK 6890

DICK 6896

WTFX
Replaces=3d6e5fdd-a0216b6d%40192.168.0.16%3Bfrom-tag%3D5517551d4feaf72do3%3B
to-tag%3DXB17S1pHp3X8a

DICK 6890

DICK 6896

WTFX
Replaces=f7ddd747-59b1035b%40192.168.0.16%3Bfrom-tag%3D717fa657bf1aff8bo3%3B
to-tag%3DjN1c6c27027Sr

DICK 6890

DICK 6896

WTFX
Replaces=e5fff425-65d08882%40192.168.0.24%3Bfrom-tag%3D716f37c97bdf792eo3%3B
to-tag%3DjcjZXKNDF7ZQg

DICK 6890

DICK 6896

WTFX
Replaces=798498b-60fe2ae9%40192.168.0.24%3Bfrom-tag%3Df86b8923d68e3031o3%3Bt
o-tag%3DHcyymrUy2a0Za

 

2012-08-04 11:13:46.491588 [DEBUG] switch_core_state_machine.c:416
(sofia/external_noauth/:

 

Anyone seen those before?

 

Sean


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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.king at quentustech.com  Sun Aug  5 07:39:54 2012
From: william.king at quentustech.com (William King)
Date: Sat, 04 Aug 2012 23:39:54 -0400
Subject: [Freeswitch-users] Odd console messages
In-Reply-To: <17cd01cd72ab$a7ec55b0$f7c50110$@bizfocused.com>
References: <165e01cd7265$be2e6f70$3a8b4e50$@bizfocused.com>
	<CANBhyPBu4RNg7vqX3TKPL4U6Ywp2e43wj7zbgG75U5o=AgxXzw@mail.gmail.com>
	<17cd01cd72ab$a7ec55b0$f7c50110$@bizfocused.com>
Message-ID: <501DEB0A.1010805@quentustech.com>

Sean,

Those lines were debug lines that a developer was using. They have since 
been removed/replaced. If you want them out of your logs then update.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 08/04/2012 09:43 PM, Sean Devoy wrote:
>
> Here is the complete log of that call.  If you want the whole 5MB say so.
>
> http://pastebin.freeswitch.com/19639
>
> I mean people call me that all the time, but Freeswitch never has!
>
> Thanks,
>
> Sean
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Anton Kvashenkin
> *Sent:* Saturday, August 04, 2012 4:05 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Odd console messages
>
> It's quite funny. Can you please post all debug log to pastebin.
>
> 04.08.2012 21:24 ???????????? "Sean Devoy" <sdevoy at bizfocused.com 
> <mailto:sdevoy at bizfocused.com>> ???????:
>
> HI All,
>
> Saw something new today and I am kind of worried that something has 
> been hacked.
>
> What does this mean in the console, it is during an incoming fax to rxfax:
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=49ed5a45-dcb3bb11%40192.168.0.24%3Bfrom-tag%3D3e5df0b5191e9761o3%3Bto-tag%3Djac81BFj6rF7e
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=7ad19169-23089a56%40192.168.0.16%3Bfrom-tag%3D82d8de4d189d12c2o3%3Bto-tag%3DN15FBmN0yyt6Q
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=200b9844-d8cf82cc%40192.168.0.24%3Bfrom-tag%3Daf559e4ebc7adaco3%3Bto-tag%3DUtBZUyQF90etQ
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=f89a0068-6396bf3c%40192.168.0.16%3Bfrom-tag%3Df9aaee7897c4286co3%3Bto-tag%3D6Bg2yD8j8jjeH
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=3d6e5fdd-a0216b6d%40192.168.0.16%3Bfrom-tag%3D5517551d4feaf72do3%3Bto-tag%3DXB17S1pHp3X8a
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=f7ddd747-59b1035b%40192.168.0.16%3Bfrom-tag%3D717fa657bf1aff8bo3%3Bto-tag%3DjN1c6c27027Sr
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=e5fff425-65d08882%40192.168.0.24%3Bfrom-tag%3D716f37c97bdf792eo3%3Bto-tag%3DjcjZXKNDF7ZQg
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=798498b-60fe2ae9%40192.168.0.24%3Bfrom-tag%3Df86b8923d68e3031o3%3Bto-tag%3DHcyymrUy2a0Za
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=49ed5a45-dcb3bb11%40192.168.0.24%3Bfrom-tag%3D3e5df0b5191e9761o3%3Bto-tag%3Djac81BFj6rF7e
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=7ad19169-23089a56%40192.168.0.16%3Bfrom-tag%3D82d8de4d189d12c2o3%3Bto-tag%3DN15FBmN0yyt6Q
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=200b9844-d8cf82cc%40192.168.0.24%3Bfrom-tag%3Daf559e4ebc7adaco3%3Bto-tag%3DUtBZUyQF90etQ
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=f89a0068-6396bf3c%40192.168.0.16%3Bfrom-tag%3Df9aaee7897c4286co3%3Bto-tag%3D6Bg2yD8j8jjeH
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=3d6e5fdd-a0216b6d%40192.168.0.16%3Bfrom-tag%3D5517551d4feaf72do3%3Bto-tag%3DXB17S1pHp3X8a
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=f7ddd747-59b1035b%40192.168.0.16%3Bfrom-tag%3D717fa657bf1aff8bo3%3Bto-tag%3DjN1c6c27027Sr
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=e5fff425-65d08882%40192.168.0.24%3Bfrom-tag%3D716f37c97bdf792eo3%3Bto-tag%3DjcjZXKNDF7ZQg
>
> DICK 6890
>
> DICK 6896
>
> WTFX 
> Replaces=798498b-60fe2ae9%40192.168.0.24%3Bfrom-tag%3Df86b8923d68e3031o3%3Bto-tag%3DHcyymrUy2a0Za
>
> 2012-08-04 11:13:46.491588 [DEBUG] switch_core_state_machine.c:416 
> (sofia/external_noauth/...
>
> Anyone seen those before?
>
> Sean
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.jugatsu at gmail.com  Sun Aug  5 09:31:52 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Sun, 5 Aug 2012 09:31:52 +0400
Subject: [Freeswitch-users] sip external port
In-Reply-To: <CAKnRR0YB92Q+w0WQQeTSX+WKiiAQD7Nr6vgiXSYz0UbBNJUcJQ@mail.gmail.com>
References: <CAKnRR0YB92Q+w0WQQeTSX+WKiiAQD7Nr6vgiXSYz0UbBNJUcJQ@mail.gmail.com>
Message-ID: <CANBhyPDRKopOT5TUoMmtPZGnfNDhEnXj+qBvpBW4d8MgfAGNMQ@mail.gmail.com>

Here comes the true power of FreeSWITCH. mod_sofia allows you to bind
different user-agents with different configuration options and therefore
different ports to be listening to. So, my question would be: why don't you
want to use 5080 port for communicate with external sip-server. Let me
explain: when freeswitch acts like UAC ( user-agent client) to register to
ITSP to get the invites for incoming calls, actually, you will be using
gateway with provided credentialf for _external_ profile and therefore all
invites will come to 5080 port, it's just like your any VoIP- client like
Zoiper, for example. It can be 23000, it doesn't matter, you even do not
need to make any port-forwardind on router ( when your PBX is behind NAT).
Speaking about NAT, another reason to use external gataway is ext-rtp and
ext-sip options @ external.xml.

But if you want that freeswitch would be acting like Registrar ( register
roadwarriors clients, for example) or/and you want to receive anonymous
calls like sip:sales at domain.ltd you acctually shoul use external profile (
i. e. 5080 port) or create another profile with port of your choise. SRV
records become very handy.

So, my point is that the name of profiles is just the name, that's all. The
configuration options does matter.

Would you want to authenticate all incoming calls - auth_calls=true/false,
etc. It was example of difference beetween internal and external profile.

Of course if you have public IP you can simply use one profile (internal
5060) for all purpose, but if you use DID from ITSP you should'n
authenticate incoming invites from that ITSP, so here comes ACL.

Sent from my cellphone,  sorry for mistakes.
05.08.2012 2:14 ???????????? "budi wibowo" <bwibowo at gmail.com> ???????:

> hi
> just curious why freeswitch using port 5080 instead of 5060 for
> communication to external sip server.
> few days ago i interconnect fs to external sip server that running on port
> 5060, and this port id fixed.
> how to overcome this situation? other pbx like asterisk can use port 5060
> for both internal and external
>
>
> regards
>
> budi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From covici at ccs.covici.com  Sun Aug  5 10:22:27 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 05 Aug 2012 02:22:27 -0400
Subject: [Freeswitch-users] sip external port
In-Reply-To: <CAMWiMy8zSzOrvH8FcZmAybHgSoYhA_vLyxtFnPLyJmnE+M3SnQ@mail.gmail.com>
References: <CAKnRR0YB92Q+w0WQQeTSX+WKiiAQD7Nr6vgiXSYz0UbBNJUcJQ@mail.gmail.com>
	<CAMWiMy8zSzOrvH8FcZmAybHgSoYhA_vLyxtFnPLyJmnE+M3SnQ@mail.gmail.com>
Message-ID: <26290.1344147747@ccs.covici.com>

You can use 5060 if you want, see the end of the file public.xml in your
dialplan directory for more information.  If you uncomment what they
say, why is 5060 any less secure?


curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> FreeSWITCH uses port 5080 for internal phones to register for security
> reasons. Although 5060 is the default port I'd advise you to leave it as is
> for obvious reasons.
> On Aug 4, 2012 3:14 PM, "budi wibowo" <bwibowo at gmail.com> wrote:
> 
> > hi
> > just curious why freeswitch using port 5080 instead of 5060 for
> > communication to external sip server.
> > few days ago i interconnect fs to external sip server that running on port
> > 5060, and this port id fixed.
> > how to overcome this situation? other pbx like asterisk can use port 5060
> > for both internal and external
> >
> >
> > regards
> >
> > budi
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 james.bravo at redmatter.com  Sun Aug  5 10:23:43 2012
From: james.bravo at redmatter.com (James Bravo)
Date: Sun, 5 Aug 2012 07:23:43 +0100
Subject: [Freeswitch-users] Generic headers in CUSTOM events
Message-ID: <501E116F.3000501@redmatter.com>

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From william.king at quentustech.com  Sun Aug  5 10:53:12 2012
From: william.king at quentustech.com (William King)
Date: Sun, 05 Aug 2012 02:53:12 -0400
Subject: [Freeswitch-users] sip external port
In-Reply-To: <26290.1344147747@ccs.covici.com>
References: <CAKnRR0YB92Q+w0WQQeTSX+WKiiAQD7Nr6vgiXSYz0UbBNJUcJQ@mail.gmail.com>
	<CAMWiMy8zSzOrvH8FcZmAybHgSoYhA_vLyxtFnPLyJmnE+M3SnQ@mail.gmail.com>
	<26290.1344147747@ccs.covici.com>
Message-ID: <501E1858.8020804@quentustech.com>

Generally 5060 is less secure because that is the port that is commonly 
scanned for by most automated scanners. So using the (semi) non standard 
port of 5080 tends to remove a lot of that headache.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 08/05/2012 02:22 AM, covici at ccs.covici.com wrote:
> You can use 5060 if you want, see the end of the file public.xml in your
> dialplan directory for more information.  If you uncomment what they
> say, why is 5060 any less secure?
>
>
> curriegrad2004<curriegrad2004 at gmail.com>  wrote:
>
>> FreeSWITCH uses port 5080 for internal phones to register for security
>> reasons. Although 5060 is the default port I'd advise you to leave it as is
>> for obvious reasons.
>> On Aug 4, 2012 3:14 PM, "budi wibowo"<bwibowo at gmail.com>  wrote:
>>
>>> hi
>>> just curious why freeswitch using port 5080 instead of 5060 for
>>> communication to external sip server.
>>> few days ago i interconnect fs to external sip server that running on port
>>> 5060, and this port id fixed.
>>> how to overcome this situation? other pbx like asterisk can use port 5060
>>> for both internal and external
>>>
>>>
>>> regards
>>>
>>> budi
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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:
>>
>> ----------------------------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org


From errotan at elder.hu  Sun Aug  5 13:45:23 2012
From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=)
Date: Sun, 05 Aug 2012 11:45:23 +0200
Subject: [Freeswitch-users] CUSTOM event messages missing
	FreeSWITCH-Hostname
In-Reply-To: <501BC9F9.3070703@redmatter.com>
References: <501BC9F9.3070703@redmatter.com>
Message-ID: <501E40B3.7030106@elder.hu>

Hi.

I have created a Jira and a patch for you: 
http://jira.freeswitch.org/browse/FS-4499

Check it out :)


2012-08-03 14:54 keltez?ssel, James Bravo ?rta:
> Hi
>
> We've recently upgraded from version 1.0.head to 1.2.0-rc2
> and noticed that certain events sent from php code using
> the ESP library have certain core headers missing, in
> particular 'FreeSWITCH-Hostname'.
> For example, using '/events plain all' in fs_cli,
> version 1.0.head would show
>
>     RECV EVENT
>     Event-Subclass: rmCTI::TRANSFER_ANSWER
>     Core-UUID: 47527f1f-b6c3-4f74-9f16-32f565ee4123
>     FreeSWITCH-Hostname: s01pbx01.test.com
>     FreeSWITCH-Switchname: s01pbx01.test.com
>     FreeSWITCH-IPv4: 211.141.141.131
>     FreeSWITCH-IPv6: ::1
>     Event-Date-Local: 2012-08-03 10:43:06
>     Event-Date-GMT: Fri, 03 Aug 2012 09:43:06 GMT
>     Event-Date-Timestamp: 1343986986238409
>     Event-Calling-File: mod_event_socket.c
>     Event-Calling-Function: read_packet
>     Event-Calling-Line-Number: 1188
>     Command: sendevent CUSTOM
>     Event-Name: CUSTOM
>     Channel-Name: sofia/internal1/1202 at sip.test.com
>     rmCTI-UUID: aebc98dc-22da-4ad1-8234-912487329123
>     rmCTI-Other-UUID: b24a8783-16b1-495e-9c4e-5ea52c362123
>     rmCTI-Info: Transfer to new b-leg answered
>
> where as version 1.2.0-rc2 it shows
>
>     RECV EVENT
>     Event-Subclass: rmCTI::TRANSFER_ANSWER
>     Command: sendevent CUSTOM
>     Event-Name: CUSTOM
>     Channel-Name: sofia/internal1/1202 at sip.test.com
>     rmCTI-UUID: c3c7b331-0b8b-4ea5-834f-9d72fd446123
>     rmCTI-Other-UUID: db75c60b-a258-4b47-8d51-a1b7ac86f123
>     rmCTI-Info: Transfer to new b-leg answered
>     Event-UUID: f47597b2-9e8c-4508-be57-3e1eed35c123
>
> Is there any reason these core headers are missing?
> Is it something to do with CUSTOM events?
>
> Thanks in advance
> James Bravo
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From james.bravo at redmatter.com  Sun Aug  5 15:44:46 2012
From: james.bravo at redmatter.com (James Bravo)
Date: Sun, 5 Aug 2012 12:44:46 +0100
Subject: [Freeswitch-users] CUSTOM event messages missing
	FreeSWITCH-Hostname
In-Reply-To: <501E40B3.7030106@elder.hu>
References: <501BC9F9.3070703@redmatter.com> <501E40B3.7030106@elder.hu>
Message-ID: <501E5CAE.6080305@redmatter.com>

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From ocset at the800group.com  Sun Aug  5 19:10:06 2012
From: ocset at the800group.com (ocset)
Date: Sun, 05 Aug 2012 23:10:06 +0800
Subject: [Freeswitch-users] Implications of using -nonat
Message-ID: <501E8CCE.5060209@the800group.com>

Hi

What are the implications of using the -nonat flag when starting 
FreeSWITCH? When should it be used?

I have started FreeSWITCH with -nonat and it registers with my external 
VOIP gateway (Pennytel) and users can make calls using the external sip 
profile. Clients outside of my network are not able to see or register 
with FreeSWITCH which is expected behaviour.

What am I missing in terms of functionality because of disabling NAT?

Thanks in advance
O.


From lazyvirus at gmx.com  Sun Aug  5 19:53:38 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Sun, 5 Aug 2012 17:53:38 +0200
Subject: [Freeswitch-users] Implications of using -nonat
In-Reply-To: <501E8CCE.5060209@the800group.com>
References: <501E8CCE.5060209@the800group.com>
Message-ID: <20120805175338.704d62bd@anubis.defcon1>

On Sun, 05 Aug 2012 23:10:06 +0800
ocset <ocset at the800group.com> wrote:

> What are the implications of using the -nonat flag when starting 
> FreeSWITCH? When should it be used?
> 
> I have started FreeSWITCH with -nonat and it registers with my
> external VOIP gateway (Pennytel) and users can make calls using
> the external sip profile. Clients outside of my network are not
> able to see or register with FreeSWITCH which is expected
> behaviour.
> 
> What am I missing in terms of functionality because of disabling
> NAT?

I think you have the answer in the question; if your server is
directly connected to the Internet AND it is also known as your
Internet address, then 'nonat' is logic, but if it is not the
case (behind a FW&|router for example) it must know its external
address as well (so no 'nonat':)

JY
-- 
Tommy dit : are you clear for tomorrow Aristote homework?
Manus dit : Yap, just dicovrd he's dead
Tommy dit : ...


From curriegrad2004 at gmail.com  Sun Aug  5 20:41:32 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 5 Aug 2012 09:41:32 -0700
Subject: [Freeswitch-users] Implications of using -nonat
In-Reply-To: <20120805175338.704d62bd@anubis.defcon1>
References: <501E8CCE.5060209@the800group.com>
	<20120805175338.704d62bd@anubis.defcon1>
Message-ID: <CAMWiMy8zbxVKc+agMzMOAuSvTo9TcELMz7nydG=WxLTf0TJ2JQ@mail.gmail.com>

There aren't any implications. The -nonat switch just simply tells
FreeSWITCH to discover for an UPnP enabled NAT gateway so it can
forward ports dynamically via that API. If you choose to open up ports
the manual way then having that -nonat switch enabled doesn't make
much of a difference.

What is important however is that you set up your ext_rtp_ip and
ext_sip_ip's correctly in your Sofia SIP profile. Examples of a very
good value in ext_rtp_ip and ext_sip_ip is stun:stun.freeswitch.org


On Sun, Aug 5, 2012 at 8:53 AM, Bzzz <lazyvirus at gmx.com> wrote:
> On Sun, 05 Aug 2012 23:10:06 +0800
> ocset <ocset at the800group.com> wrote:
>
>> What are the implications of using the -nonat flag when starting
>> FreeSWITCH? When should it be used?
>>
>> I have started FreeSWITCH with -nonat and it registers with my
>> external VOIP gateway (Pennytel) and users can make calls using
>> the external sip profile. Clients outside of my network are not
>> able to see or register with FreeSWITCH which is expected
>> behaviour.
>>
>> What am I missing in terms of functionality because of disabling
>> NAT?
>
> I think you have the answer in the question; if your server is
> directly connected to the Internet AND it is also known as your
> Internet address, then 'nonat' is logic, but if it is not the
> case (behind a FW&|router for example) it must know its external
> address as well (so no 'nonat':)
>
> JY
> --
> Tommy dit : are you clear for tomorrow Aristote homework?
> Manus dit : Yap, just dicovrd he's dead
> Tommy dit : ...
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From lazyvirus at gmx.com  Sun Aug  5 20:56:07 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Sun, 5 Aug 2012 18:56:07 +0200
Subject: [Freeswitch-users] Implications of using -nonat
In-Reply-To: <CAMWiMy8zbxVKc+agMzMOAuSvTo9TcELMz7nydG=WxLTf0TJ2JQ@mail.gmail.com>
References: <501E8CCE.5060209@the800group.com>
	<20120805175338.704d62bd@anubis.defcon1>
	<CAMWiMy8zbxVKc+agMzMOAuSvTo9TcELMz7nydG=WxLTf0TJ2JQ@mail.gmail.com>
Message-ID: <20120805185607.08a9797a@anubis.defcon1>

On Sun, 5 Aug 2012 09:41:32 -0700
curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> There aren't any implications. The -nonat switch just simply tells
> FreeSWITCH to discover for an UPnP enabled NAT gateway so it can
> forward ports dynamically via that API. If you choose to open up
> ports the manual way then having that -nonat switch enabled
> doesn't make much of a difference.

My bad: too much time w/o touching FS (but very bad name too,
noupnpnat would have been a lot more logical:( 

JY
-- 
<Strasph> If I die in 5 minutes, what do you wanna tell me ?
<Decibel> "What's your root password?"


From curriegrad2004 at gmail.com  Sun Aug  5 21:18:28 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 5 Aug 2012 10:18:28 -0700
Subject: [Freeswitch-users] Implications of using -nonat
In-Reply-To: <20120805185607.08a9797a@anubis.defcon1>
References: <501E8CCE.5060209@the800group.com>
	<20120805175338.704d62bd@anubis.defcon1>
	<CAMWiMy8zbxVKc+agMzMOAuSvTo9TcELMz7nydG=WxLTf0TJ2JQ@mail.gmail.com>
	<20120805185607.08a9797a@anubis.defcon1>
Message-ID: <CAMWiMy_aSWdtVT6vQXLiMaHnr2Cuy=DktrP7GmT3innJOFJXUg@mail.gmail.com>

I've submitted a patch for that :P

On Sun, Aug 5, 2012 at 9:56 AM, Bzzz <lazyvirus at gmx.com> wrote:
> On Sun, 5 Aug 2012 09:41:32 -0700
> curriegrad2004 <curriegrad2004 at gmail.com> wrote:
>
>> There aren't any implications. The -nonat switch just simply tells
>> FreeSWITCH to discover for an UPnP enabled NAT gateway so it can
>> forward ports dynamically via that API. If you choose to open up
>> ports the manual way then having that -nonat switch enabled
>> doesn't make much of a difference.
>
> My bad: too much time w/o touching FS (but very bad name too,
> noupnpnat would have been a lot more logical:(
>
> JY
> --
> <Strasph> If I die in 5 minutes, what do you wanna tell me ?
> <Decibel> "What's your root password?"
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From gmaruzz at gmail.com  Sun Aug  5 23:00:17 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 5 Aug 2012 21:00:17 +0200
Subject: [Freeswitch-users] Room Sharing Offer at Cluecon Hyatt
Message-ID: <CALXCt0o6+31U9B1bX4TucHZixCr65jbCZz-q66maAZwSc+yMYg@mail.gmail.com>

Hi FreeSWITCHers,

seems some of Cluecon participants have not find a room at Hyatt.

I can share a room with two queen size beds.

If you're interested, contact me at the cellphone in signature (sms
ok), or private mail at gmaruzz at gmail.com .

Don't forget to give me your phone number too.

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From bwibowo at gmail.com  Mon Aug  6 01:50:04 2012
From: bwibowo at gmail.com (budi wibowo)
Date: Mon, 6 Aug 2012 04:50:04 +0700
Subject: [Freeswitch-users] sip external port
In-Reply-To: <501E1858.8020804@quentustech.com>
References: <CAKnRR0YB92Q+w0WQQeTSX+WKiiAQD7Nr6vgiXSYz0UbBNJUcJQ@mail.gmail.com>
	<CAMWiMy8zSzOrvH8FcZmAybHgSoYhA_vLyxtFnPLyJmnE+M3SnQ@mail.gmail.com>
	<26290.1344147747@ccs.covici.com>
	<501E1858.8020804@quentustech.com>
Message-ID: <CAKnRR0ay-yetumTXH6Ta6ox_Gb4QjcG+ES-zOv-WOF4UiAnmyg@mail.gmail.com>

actually agree, but for some case i have partner that only can use 5060,
not support binding multiple port also :)
anyway thx for the reasonable explanation

br

budi


On Sun, Aug 5, 2012 at 1:53 PM, William King
<william.king at quentustech.com>wrote:

> Generally 5060 is less secure because that is the port that is commonly
> scanned for by most automated scanners. So using the (semi) non standard
> port of 5080 tends to remove a lot of that headache.
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
>
> On 08/05/2012 02:22 AM, covici at ccs.covici.com wrote:
> > You can use 5060 if you want, see the end of the file public.xml in your
> > dialplan directory for more information.  If you uncomment what they
> > say, why is 5060 any less secure?
> >
> >
> > curriegrad2004<curriegrad2004 at gmail.com>  wrote:
> >
> >> FreeSWITCH uses port 5080 for internal phones to register for security
> >> reasons. Although 5060 is the default port I'd advise you to leave it
> as is
> >> for obvious reasons.
> >> On Aug 4, 2012 3:14 PM, "budi wibowo"<bwibowo at gmail.com>  wrote:
> >>
> >>> hi
> >>> just curious why freeswitch using port 5080 instead of 5060 for
> >>> communication to external sip server.
> >>> few days ago i interconnect fs to external sip server that running on
> port
> >>> 5060, and this port id fixed.
> >>> how to overcome this situation? other pbx like asterisk can use port
> 5060
> >>> for both internal and external
> >>>
> >>>
> >>> regards
> >>>
> >>> budi
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> Join Us At ClueCon - Aug 7-9, 2012
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at 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:
> >>
> >> ----------------------------------------------------
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Aug  6 08:38:34 2012
From: jack at livecall.com (Jack)
Date: Sun, 05 Aug 2012 21:38:34 -0700
Subject: [Freeswitch-users] RTMP Client not responding in Chrome
Message-ID: <501F4A4A.9040307@livecall.com>

Is anyone else experiencing problems with RTMP FLEX client in Chrome?
Just today it started making chrome become unresponsive.  If I allow it 
to wait about 3 minutes, sometimes it finally starts working.
I am experiencing this on my site as well as the conference.freeswitch 
.org/conf site.
It is working fine in IE Safari, and Firefox.
jack


From kochanowski.wojtek at gmail.com  Mon Aug  6 12:42:54 2012
From: kochanowski.wojtek at gmail.com (Wojtek Kochanowski)
Date: Mon, 6 Aug 2012 10:42:54 +0200
Subject: [Freeswitch-users] RTMP Client not responding in Chrome
In-Reply-To: <501F4A4A.9040307@livecall.com>
References: <501F4A4A.9040307@livecall.com>
Message-ID: <CAKDokcAo89Az=eekfCEEJZb6bgpVanGrh3Y0hdVSc8FdP6jd1A@mail.gmail.com>

Indeed, rtmp client doesn't work for me on Chrome/Chromium also.
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From vbvbrj at gmail.com  Mon Aug  6 12:56:03 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Mon, 06 Aug 2012 11:56:03 +0300
Subject: [Freeswitch-users] Callcenter saying position in queue
Message-ID: <501F86A3.4060904@gmail.com>

Hello.
I want to configure callcenter in so way, that every member in queue 
will hear periodically its position in queue. This may be done using lua 
script and using uuid_broadcast. But how in lua script get the members 
position? From callcenter_config queue list members [queue_name] cant 
find any position. Even base_score is always 0 for all members using 
<param name="time-base-score" value="system"/> in callcenter configuration.
Thank you.


From odermann at googlemail.com  Mon Aug  6 13:26:52 2012
From: odermann at googlemail.com (Dennis)
Date: Mon, 6 Aug 2012 11:26:52 +0200
Subject: [Freeswitch-users] Fax (RX): Missing name/header variable in
	socket!?
In-Reply-To: <501BE1E6.1030101@coppice.org>
References: <CALaOL+hYAxkVTEOLBCpCZB5H6LbZaLH_hW-UiagvP_XYAKOQbQ@mail.gmail.com>
	<501BB72D.7070106@coppice.org>
	<CALaOL+h7-6pTtf0rVHD5X0wryawhyn7UZ5v87-mZ-jwXinE_gg@mail.gmail.com>
	<501BE1E6.1030101@coppice.org>
Message-ID: <CALaOL+j3+pCk1x5=Wj6+sbDgyKeHW=eUZtCT4j-1zH+W_pJeBw@mail.gmail.com>

thanks, steve, then we will have to find something else.

kind regards
dennis


From odermann at googlemail.com  Mon Aug  6 13:35:07 2012
From: odermann at googlemail.com (Dennis)
Date: Mon, 6 Aug 2012 11:35:07 +0200
Subject: [Freeswitch-users] Early media: Sending 180 instead of 183?
In-Reply-To: <CALaOL+gnBq-L8Nu3W76QJchzgYbw59BKGF44oWrzRbt0GDX7CA@mail.gmail.com>
References: <CALaOL+gnBq-L8Nu3W76QJchzgYbw59BKGF44oWrzRbt0GDX7CA@mail.gmail.com>
Message-ID: <CALaOL+jXbbz6YxT8pi3zchSa0gtnXCpyrFkq=kHzPLMedxb+rQ@mail.gmail.com>

we know about ring_ready and we played with it, but as you can see in
the screenshot (https://www.dropbox.com/s/4ffz2vasvgct0j6/fs.png), fs
automatically sends a 183 message (directly after the 180). most of
the time this is ok, but in germany it seems to make problems.

in our case we would like fs NOT to send the 183 automatically - or we
would like to see a way, to tell fs not to send the 183 in some cases.
we tested this with different carriers and we always have the same
problems.

kind regards
dennis


From peter.olsson at visionutveckling.se  Mon Aug  6 13:43:52 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 6 Aug 2012 09:43:52 +0000
Subject: [Freeswitch-users] Early media: Sending 180 instead of 183?
Message-ID: <1FFF97C269757C458224B7C895F35F1513DF6F@cantor.std.visionutv.se>

If you don't try to do some kind of playback before the call is answered, no early media will be sent. Also make sure you don't set ringback or transfer_ringback variables, and if you're doing a bridge, make sure to set ignore_early_media=true.

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Dennis
Skickat: den 6 augusti 2012 11:35
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Early media: Sending 180 instead of 183?

we know about ring_ready and we played with it, but as you can see in the screenshot (https://www.dropbox.com/s/4ffz2vasvgct0j6/fs.png), fs automatically sends a 183 message (directly after the 180). most of the time this is ok, but in germany it seems to make problems.

in our case we would like fs NOT to send the 183 automatically - or we would like to see a way, to tell fs not to send the 183 in some cases.
we tested this with different carriers and we always have the same problems.

kind regards
dennis

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at 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:501f8da832768810110182!



From avi at avimarcus.net  Mon Aug  6 13:49:57 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 6 Aug 2012 12:49:57 +0300
Subject: [Freeswitch-users] Early media: Sending 180 instead of 183?
In-Reply-To: <CALaOL+jXbbz6YxT8pi3zchSa0gtnXCpyrFkq=kHzPLMedxb+rQ@mail.gmail.com>
References: <CALaOL+gnBq-L8Nu3W76QJchzgYbw59BKGF44oWrzRbt0GDX7CA@mail.gmail.com>
	<CALaOL+jXbbz6YxT8pi3zchSa0gtnXCpyrFkq=kHzPLMedxb+rQ@mail.gmail.com>
Message-ID: <CALNZuw=hYVuHF6vapL9RsjwNE9wjGZM1HL0VgXa=j9GPXfRDPw@mail.gmail.com>

Maybe you need ignore_early_media to not pass along 183 from your
endpoint...?

-Avi


On Mon, Aug 6, 2012 at 12:35 PM, Dennis <odermann at googlemail.com> wrote:

> we know about ring_ready and we played with it, but as you can see in
> the screenshot (https://www.dropbox.com/s/4ffz2vasvgct0j6/fs.png), fs
> automatically sends a 183 message (directly after the 180). most of
> the time this is ok, but in germany it seems to make problems.
>
> in our case we would like fs NOT to send the 183 automatically - or we
> would like to see a way, to tell fs not to send the 183 in some cases.
> we tested this with different carriers and we always have the same
> problems.
>
> kind regards
> dennis
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From admin at blindi.net  Mon Aug  6 13:53:43 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Mon, 6 Aug 2012 11:53:43 +0200 (CEST)
Subject: [Freeswitch-users] question say a none-us standard date and time
	format?
In-Reply-To: <CALaOL+jXbbz6YxT8pi3zchSa0gtnXCpyrFkq=kHzPLMedxb+rQ@mail.gmail.com>
References: <CALaOL+gnBq-L8Nu3W76QJchzgYbw59BKGF44oWrzRbt0GDX7CA@mail.gmail.com>
	<CALaOL+jXbbz6YxT8pi3zchSa0gtnXCpyrFkq=kHzPLMedxb+rQ@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1208061144060.31613@1a-8140.antagus.de>

Hi guys,
how can i say a none-us standard time and date please?
I.m. in germany.
the german date format is:
day, month, year, hour, minute, seconds.
how can i change this format please?
or how can i say a timestamp in this format from a variable please?
I don.t find a feature to change this format.

Can your help please?
thanks.




---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
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http://www.blindi.net/mailman/listinfo/blinde-misc

From odermann at googlemail.com  Mon Aug  6 14:19:09 2012
From: odermann at googlemail.com (Dennis)
Date: Mon, 6 Aug 2012 12:19:09 +0200
Subject: [Freeswitch-users] Early media: Sending 180 instead of 183?
In-Reply-To: <CALNZuw=hYVuHF6vapL9RsjwNE9wjGZM1HL0VgXa=j9GPXfRDPw@mail.gmail.com>
References: <CALaOL+gnBq-L8Nu3W76QJchzgYbw59BKGF44oWrzRbt0GDX7CA@mail.gmail.com>
	<CALaOL+jXbbz6YxT8pi3zchSa0gtnXCpyrFkq=kHzPLMedxb+rQ@mail.gmail.com>
	<CALNZuw=hYVuHF6vapL9RsjwNE9wjGZM1HL0VgXa=j9GPXfRDPw@mail.gmail.com>
Message-ID: <CALaOL+h8xSM19i+DmeNhg7vj35s9+pXwwcsGZ=5NPPs7Gq7kpg@mail.gmail.com>

in this special case, we do NOT have an outgoing channel and we do
ALWAYS use ignore_early_media, if we have an outgoing channel....

we do/want the following: incoming call -> playback audio to caller ->
hangup (no bridge or anything else)

out test-dialplan is as follows:

<extension name="test">
 ?<condition field="destination_number" expression="^01234567$">
 ? <action application="ring_ready"/>
 ? <action application="set" data="ignore_early_media=true"/> // with,
ans without this line
 ? <action application="playback" data="/tmp/test.wav"/>
 ?</condition>
</extension>

everything would work with only the 180 - as soon as the 183 is sent,
problems can come up. the 183 is simply to much in this case.


From peter.olsson at visionutveckling.se  Mon Aug  6 14:27:03 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 6 Aug 2012 10:27:03 +0000
Subject: [Freeswitch-users] Early media: Sending 180 instead of 183?
Message-ID: <1FFF97C269757C458224B7C895F35F1513E1A1@cantor.std.visionutv.se>

If you do playback before you answer the call - that IS early media. If you answer the call before playback it will not send 183. But if you want to play it using early media it's just not possible to play it without using 183.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Dennis
Skickat: den 6 augusti 2012 12:19
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Early media: Sending 180 instead of 183?

in this special case, we do NOT have an outgoing channel and we do ALWAYS use ignore_early_media, if we have an outgoing channel....

we do/want the following: incoming call -> playback audio to caller -> hangup (no bridge or anything else)

out test-dialplan is as follows:

<extension name="test">
 ?<condition field="destination_number" expression="^01234567$">
 ? <action application="ring_ready"/>
 ? <action application="set" data="ignore_early_media=true"/> // with, ans without this line
 ? <action application="playback" data="/tmp/test.wav"/>
 ?</condition>
</extension>

everything would work with only the 180 - as soon as the 183 is sent, problems can come up. the 183 is simply to much in this case.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at 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:501f980b32761254510098!



From tonybecq at yahoo.fr  Mon Aug  6 14:46:31 2012
From: tonybecq at yahoo.fr (8hector8)
Date: Mon, 6 Aug 2012 03:46:31 -0700 (PDT)
Subject: [Freeswitch-users]  simulating a modem transaction...
Message-ID: <34260552.post@talk.nabble.com>


Hello everybody,

Here is my issue. I need to simulate a modem transaction in C. It's a V.21
modem in 200 bits/s. This modem dial a number and waits for an "request to
send" sent by the reciever. This is where I have to simulate a modem.
The "request to send" is composed of a 2100 Hz tone for 2,4 secondes and 32
alternations of two tones :
1650 Hz and 1850 Hz, 15 ms each.
In my idea, streamFile() function seams to be the good mean.

The real issue is that when those tones are sent, the transmeter will send a
"message" which is 16 digits in V.21 mode. 8 data bits, no parity, 2 stop
bits.
200 bits/s means that each tone is 5 ms. 1180 Hz is a "1" and 980 is a "0".

So, I need to get those 16 digits messages from tones back into numbers. So
I need a way to listen the signals and analyse them and this is where I
don't know how to do it.

Maybe, I have to create a module into freeswitch and here also I need an
help to do it.

Thanks for your attention.

Tony BECQ
-- 
View this message in context: http://old.nabble.com/simulating-a-modem-transaction...-tp34260552p34260552.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From odermann at googlemail.com  Mon Aug  6 14:58:19 2012
From: odermann at googlemail.com (Dennis)
Date: Mon, 6 Aug 2012 12:58:19 +0200
Subject: [Freeswitch-users] Early media: Sending 180 instead of 183?
In-Reply-To: <1FFF97C269757C458224B7C895F35F1513E1A1@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1513E1A1@cantor.std.visionutv.se>
Message-ID: <CALaOL+g_jpOZ1T0cAqB2rKACxKFT8u2Od8L9HRYz_2O9GG7mDA@mail.gmail.com>

we are doing a lot with ISDN-connections, while ISDN in germany seems
to be very different from the ISDN in the usa.

our carriers asked us to only send a 180 instead of the 183 because,
as i wrote above, 183 messages are not mapped reliable to ISUP as
written in the specification ITU-T Rec. Q.1912.5.
the a-leg sends an IAM or INVITE, but does not receive an answer -
this can differ from a-leg to a-leg.

we know, that this problem is quite special, but we have the same
problems with different carriers.


From peter.olsson at visionutveckling.se  Mon Aug  6 15:34:23 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 6 Aug 2012 11:34:23 +0000
Subject: [Freeswitch-users] Early media: Sending 180 instead of 183?
Message-ID: <1FFF97C269757C458224B7C895F35F1513E262@cantor.std.visionutv.se>

If you don't want to send 183, don't try to send early media. It's the only way.

In the dialplan you posted, remove the line: <action application="playback" data="/tmp/test.wav"/>

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Dennis
Skickat: den 6 augusti 2012 12:58
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Early media: Sending 180 instead of 183?

we are doing a lot with ISDN-connections, while ISDN in germany seems to be very different from the ISDN in the usa.

our carriers asked us to only send a 180 instead of the 183 because, as i wrote above, 183 messages are not mapped reliable to ISUP as written in the specification ITU-T Rec. Q.1912.5.
the a-leg sends an IAM or INVITE, but does not receive an answer - this can differ from a-leg to a-leg.

we know, that this problem is quite special, but we have the same problems with different carriers.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at 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:501fa15e32761376919998!



From B.Tietz at pinguin.ag  Mon Aug  6 16:00:07 2012
From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag)
Date: Mon, 6 Aug 2012 14:00:07 +0200
Subject: [Freeswitch-users] tel uri
Message-ID: <07BF4904977CC645B485E970424193AD118F3F82F0@localhost>

Hi,

is there a way to send a tel uri instead of a sip uri in an initial INVITE!?

so I need:

FROM: <tel:+441234567>

and not 

FROM: <sip:00441234567 at domain.sip>

regards,
Benjamin




From B.Tietz at pinguin.ag  Mon Aug  6 16:46:23 2012
From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag)
Date: Mon, 6 Aug 2012 14:46:23 +0200
Subject: [Freeswitch-users] tel uri
Message-ID: <07BF4904977CC645B485E970424193AD118F4FBB4C@localhost>

Hi,

as mentioned from IRC:

to quote mod_sofia:  SWITCH_LOG_ERROR, "URL Error! tel: uri's not supported at this time

if there is someone who has an idea, just write :-)

regards,
Benjamin


>Hi,
>is there a way to send a tel uri instead of a sip uri in an initial INVITE!?
>so I need:
>
>FROM: <tel:+441234567>
>and not 
>
>FROM: <sip:00441234567 at domain.sip>



From alex at thewinelake.com  Mon Aug  6 17:22:59 2012
From: alex at thewinelake.com (Alex)
Date: Mon, 06 Aug 2012 14:22:59 +0100
Subject: [Freeswitch-users] New SIP profile
In-Reply-To: <501BD835.3060308@bestnet.kharkov.ua>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>	<50178DB7.7000606@bestnet.kharkov.ua>	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>	<50194496.10804@bestnet.kharkov.ua>	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>	<501A5AF4.9020008@bestnet.kharkov.ua>	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>	<501A6085.2020308@bestnet.kharkov.ua>	<CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>	<501BAE64.3020304@bestnet.kharkov.ua>	<CAJUJwtgHFyrk7ybxAmH2xmaz=d=OgCy5uHOmGhnyOK3QpdRjRw@mail.gmail.com>	<CAPBKugskuRA2FgE6H+fUsHrU4+N-E+2erantqShSKn7H6Qw99g@mail.gmail.com>
	<CAJUJwti68vmSDFpAkwk08otnJyWHkOWJ6qRJ=AcStmtW=RT04Q@mail.gmail.com>
	<501BD835.3060308@bestnet.kharkov.ua>
Message-ID: <501FC533.60708@thewinelake.com>

How do I get Freeswitch to load in a new sip profile without restarting 
it or mod_sofia?


From steveu at coppice.org  Mon Aug  6 17:25:00 2012
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 06 Aug 2012 21:25:00 +0800
Subject: [Freeswitch-users] simulating a modem transaction...
In-Reply-To: <34260552.post@talk.nabble.com>
References: <34260552.post@talk.nabble.com>
Message-ID: <501FC5AC.6070305@coppice.org>

On 08/06/2012 06:46 PM, 8hector8 wrote:
> Hello everybody,
>
> Here is my issue. I need to simulate a modem transaction in C. It's a V.21
> modem in 200 bits/s. This modem dial a number and waits for an "request to
> send" sent by the reciever. This is where I have to simulate a modem.
> The "request to send" is composed of a 2100 Hz tone for 2,4 secondes and 32
> alternations of two tones :
> 1650 Hz and 1850 Hz, 15 ms each.
> In my idea, streamFile() function seams to be the good mean.
>
> The real issue is that when those tones are sent, the transmeter will send a
> "message" which is 16 digits in V.21 mode. 8 data bits, no parity, 2 stop
> bits.
> 200 bits/s means that each tone is 5 ms. 1180 Hz is a "1" and 980 is a "0".
>
> So, I need to get those 16 digits messages from tones back into numbers. So
> I need a way to listen the signals and analyse them and this is where I
> don't know how to do it.
>
> Maybe, I have to create a module into freeswitch and here also I need an
> help to do it.
>
> Thanks for your attention.
>
> Tony BECQ
The V.21 modem in spandsp can easily be modified to work at 200bps. 
There are setups in there for 110bps and 300bps. Just add another table 
entry to 200bps. Wrap that in some endpoint code, rather like the FAX 
modem stuff, and you're pretty much there.

Steve



From eagle.antonio at gmail.com  Mon Aug  6 17:25:57 2012
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 6 Aug 2012 13:25:57 +0000
Subject: [Freeswitch-users] New SIP profile
In-Reply-To: <501FC533.60708@thewinelake.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<50194496.10804@bestnet.kharkov.ua>
	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>
	<501A5AF4.9020008@bestnet.kharkov.ua>
	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>
	<501A6085.2020308@bestnet.kharkov.ua>
	<CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>
	<501BAE64.3020304@bestnet.kharkov.ua>
	<CAJUJwtgHFyrk7ybxAmH2xmaz=d=OgCy5uHOmGhnyOK3QpdRjRw@mail.gmail.com>
	<CAPBKugskuRA2FgE6H+fUsHrU4+N-E+2erantqShSKn7H6Qw99g@mail.gmail.com>
	<CAJUJwti68vmSDFpAkwk08otnJyWHkOWJ6qRJ=AcStmtW=RT04Q@mail.gmail.com>
	<501BD835.3060308@bestnet.kharkov.ua>
	<501FC533.60708@thewinelake.com>
Message-ID: <CAF+3j-v7LudQh6T7s7WyCOuMjnBkPHR3uD9PdP_LNc=VV7k00g@mail.gmail.com>

*http://wiki.freeswitch.org/wiki/Mod_sofia#Starting_a_new_profile*

2012/8/6 Alex <alex at thewinelake.com>

> How do I get Freeswitch to load in a new sip profile without restarting
> it or mod_sofia?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From alex at thewinelake.com  Mon Aug  6 17:32:39 2012
From: alex at thewinelake.com (Alex)
Date: Mon, 06 Aug 2012 14:32:39 +0100
Subject: [Freeswitch-users] New SIP profile
In-Reply-To: <CAF+3j-v7LudQh6T7s7WyCOuMjnBkPHR3uD9PdP_LNc=VV7k00g@mail.gmail.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<50194496.10804@bestnet.kharkov.ua>
	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>
	<501A5AF4.9020008@bestnet.kharkov.ua>
	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>
	<501A6085.2020308@bestnet.kharkov.ua>
	<CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>
	<501BAE64.3020304@bestnet.kharkov.ua>
	<CAJUJwtgHFyrk7ybxAmH2xmaz=d=OgCy5uHOmGhnyOK3QpdRjRw@mail.gmail.com>
	<CAPBKugskuRA2FgE6H+fUsHrU4+N-E+2erantqShSKn7H6Qw99g@mail.gmail.com>
	<CAJUJwti68vmSDFpAkwk08otnJyWHkOWJ6qRJ=AcStmtW=RT04Q@mail.gmail.com>
	<501BD835.3060308@bestnet.kharkov.ua>
	<501FC533.60708@thewinelake.com>
	<CAF+3j-v7LudQh6T7s7WyCOuMjnBkPHR3uD9PdP_LNc=VV7k00g@mail.gmail.com>
Message-ID: <501FC777.3070602@thewinelake.com>

Thanks - I've been playing with those commands and am seeing some 
slightly strange results.
For example, "sofia profile external rescan reloadxml" - I'm not sure 
that's doing the right thing.
I get messages like "Ignoring duplicate gateway XXXX" which seems a 
little odd.


> *http://wiki.freeswitch.org/wiki/Mod_sofia#Starting_a_new_profile*
>
> 2012/8/6 Alex <alex at thewinelake.com <mailto:alex at thewinelake.com>>
>
>     How do I get Freeswitch to load in a new sip profile without
>     restarting
>     it or mod_sofia?
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     FreeSWITCH-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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2196 / Virus Database: 2437/5180 - Release Date: 08/05/12
>

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From alex at thewinelake.com  Mon Aug  6 17:34:32 2012
From: alex at thewinelake.com (Alex)
Date: Mon, 06 Aug 2012 14:34:32 +0100
Subject: [Freeswitch-users] New SIP profile
In-Reply-To: <CAF+3j-v7LudQh6T7s7WyCOuMjnBkPHR3uD9PdP_LNc=VV7k00g@mail.gmail.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<50194496.10804@bestnet.kharkov.ua>
	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>
	<501A5AF4.9020008@bestnet.kharkov.ua>
	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>
	<501A6085.2020308@bestnet.kharkov.ua>
	<CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>
	<501BAE64.3020304@bestnet.kharkov.ua>
	<CAJUJwtgHFyrk7ybxAmH2xmaz=d=OgCy5uHOmGhnyOK3QpdRjRw@mail.gmail.com>
	<CAPBKugskuRA2FgE6H+fUsHrU4+N-E+2erantqShSKn7H6Qw99g@mail.gmail.com>
	<CAJUJwti68vmSDFpAkwk08otnJyWHkOWJ6qRJ=AcStmtW=RT04Q@mail.gmail.com>
	<501BD835.3060308@bestnet.kharkov.ua>
	<501FC533.60708@thewinelake.com>
	<CAF+3j-v7LudQh6T7s7WyCOuMjnBkPHR3uD9PdP_LNc=VV7k00g@mail.gmail.com>
Message-ID: <501FC7E8.8030609@thewinelake.com>

Also, I get the feeling that rescan does a full reloadxml even if you 
don't ask for it.
so

  sofia profile external rescan reloadxml
is equivalent to
  sofia profile external rescan

I guess I just need to do some tests!



> *http://wiki.freeswitch.org/wiki/Mod_sofia#Starting_a_new_profile*
>
> 2012/8/6 Alex <alex at thewinelake.com <mailto:alex at thewinelake.com>>
>
>     How do I get Freeswitch to load in a new sip profile without
>     restarting
>     it or mod_sofia?
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     FreeSWITCH-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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2196 / Virus Database: 2437/5180 - Release Date: 08/05/12
>
...
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From vbvbrj at gmail.com  Mon Aug  6 17:37:29 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Mon, 06 Aug 2012 16:37:29 +0300
Subject: [Freeswitch-users] Callcenter saying position in queue
In-Reply-To: <501F86A3.4060904@gmail.com>
References: <501F86A3.4060904@gmail.com>
Message-ID: <501FC899.5030601@gmail.com>

So I managed to make a script in lua which will count callers position 
by uuid and say the position. But as soon as digit is playied back to 
caller, it is disconnected from callcenter application and hangs up. I 
want to play the digit together with the moh sound and caller will 
continue to stay in queue. Something which will transmit the digit 
playing in to the a-leg.


From alex at thewinelake.com  Mon Aug  6 17:43:13 2012
From: alex at thewinelake.com (Alex)
Date: Mon, 06 Aug 2012 14:43:13 +0100
Subject: [Freeswitch-users] New SIP profile
In-Reply-To: <501FC7E8.8030609@thewinelake.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<50194496.10804@bestnet.kharkov.ua>
	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>
	<501A5AF4.9020008@bestnet.kharkov.ua>
	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>
	<501A6085.2020308@bestnet.kharkov.ua>
	<CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>
	<501BAE64.3020304@bestnet.kharkov.ua>
	<CAJUJwtgHFyrk7ybxAmH2xmaz=d=OgCy5uHOmGhnyOK3QpdRjRw@mail.gmail.com>
	<CAPBKugskuRA2FgE6H+fUsHrU4+N-E+2erantqShSKn7H6Qw99g@mail.gmail.com>
	<CAJUJwti68vmSDFpAkwk08otnJyWHkOWJ6qRJ=AcStmtW=RT04Q@mail.gmail.com>
	<501BD835.3060308@bestnet.kharkov.ua>
	<501FC533.60708@thewinelake.com>
	<CAF+3j-v7LudQh6T7s7WyCOuMjnBkPHR3uD9PdP_LNc=VV7k00g@mail.gmail.com>
	<501FC7E8.8030609@thewinelake.com>
Message-ID: <501FC9F1.7080108@thewinelake.com>

...I think it looks like I need to do a killgw on anything that may have 
changed.
> Also, I get the feeling that rescan does a full reloadxml even if you 
> don't ask for it.
> so
>   sofia profile external rescan reloadxml
> is equivalent to
>   sofia profile external rescan
>
> I guess I just need to do some tests!
>
>
>> *http://wiki.freeswitch.org/wiki/Mod_sofia#Starting_a_new_profile*
>>
>> 2012/8/6 Alex <alex at thewinelake.com <mailto:alex at thewinelake.com>>
>>
>>     How do I get Freeswitch to load in a new sip profile without
>>     restarting
>>     it or mod_sofia?
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.com
>>
>>     Join Us At ClueCon - Aug 7-9, 2012
>>
>>     FreeSWITCH-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
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> No virus found in this message.
>> Checked by AVG - www.avg.com <http://www.avg.com>
>> Version: 2012.0.2196 / Virus Database: 2437/5180 - Release Date: 08/05/12
>>
> ...
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2196 / Virus Database: 2437/5180 - Release Date: 08/05/12
>

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From alex at thewinelake.com  Mon Aug  6 18:13:09 2012
From: alex at thewinelake.com (Alex)
Date: Mon, 06 Aug 2012 15:13:09 +0100
Subject: [Freeswitch-users] One Way Audio
In-Reply-To: <501FC7E8.8030609@thewinelake.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<50194496.10804@bestnet.kharkov.ua>
	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>
	<501A5AF4.9020008@bestnet.kharkov.ua>
	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>
	<501A6085.2020308@bestnet.kharkov.ua>
	<CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>
	<501BAE64.3020304@bestnet.kharkov.ua>
	<CAJUJwtgHFyrk7ybxAmH2xmaz=d=OgCy5uHOmGhnyOK3QpdRjRw@mail.gmail.com>
	<CAPBKugskuRA2FgE6H+fUsHrU4+N-E+2erantqShSKn7H6Qw99g@mail.gmail.com>
	<CAJUJwti68vmSDFpAkwk08otnJyWHkOWJ6qRJ=AcStmtW=RT04Q@mail.gmail.com>
	<501BD835.3060308@bestnet.kharkov.ua>
	<501FC533.60708@thewinelake.com>
	<CAF+3j-v7LudQh6T7s7WyCOuMjnBkPHR3uD9PdP_LNc=VV7k00g@mail.gmail.com>
	<501FC7E8.8030609@thewinelake.com>
Message-ID: <501FD0F5.2000600@thewinelake.com>

Getting an occasional one-way audio problem.

One thing I was wondering about is this:

switch_rtp.c:3240 Auto Changing port from 86.174.119.10:16432 to 86.174.119.10:61384

Can one somehow disable this feature as part of diagnostics?






From tonybecq at yahoo.fr  Mon Aug  6 18:27:26 2012
From: tonybecq at yahoo.fr (8hector8)
Date: Mon, 6 Aug 2012 07:27:26 -0700 (PDT)
Subject: [Freeswitch-users]  simulating a modem transaction...
Message-ID: <34260552.post@talk.nabble.com>


Hello everybody,

Here is my issue. I need to simulate a modem transaction in C. It's a V.21
modem in 200 bits/s. This modem dial a number and waits for an "request to
send" sent by the reciever. This is where I have to simulate a modem.
The "request to send" is composed of a 2100 Hz tone for 2,4 secondes and 32
alternations of two tones :
1650 Hz and 1850 Hz, 15 ms each.
In my idea, streamFile() function seams to be the good mean.

The real issue is that when those tones are sent, the transmeter will send a
"message" which is 16 digits in V.21 mode. 8 data bits, no parity, 2 stop
bits.
200 bits/s means that each tone is 5 ms. 1180 Hz is a "1" and 980 is a "0".

So, I need to get those 16 digits messages from tones back into numbers. So
I need a way to listen the signals and analyse them and this is where I
don't know how to do it.

Maybe, I have to create a module into freeswitch and here also I need an
help to do it.

Thanks for your attention.

8hector8
-- 
View this message in context: http://old.nabble.com/simulating-a-modem-transaction...-tp34260552p34260552.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From tonybecq at yahoo.fr  Mon Aug  6 18:31:22 2012
From: tonybecq at yahoo.fr (Tony BECQ)
Date: Mon, 6 Aug 2012 15:31:22 +0100 (BST)
Subject: [Freeswitch-users] Simultating a modem transaction...
Message-ID: <1344263482.50338.YahooMailNeo@web171201.mail.ir2.yahoo.com>

Hello everybody, 

Here is my issue. I need to simulate a modem transaction in C. 
It's a V.21 modem in 200 bits/s. This modem dial a number and waits for 
an "request to send" sent by the reciever. This is where I have to 
simulate a modem. 
The "request to send" is composed of a 2100 Hz tone for 2,4 secondes and 32 alternations of two tones : 
1650 Hz and 1850 Hz, 15 ms each. 
In my idea, streamFile() function seams to be the good mean. 

The real issue is that when those tones are sent, the transmeter
 will send a "message" which is 16 digits in V.21 mode. 8 data bits, no 
parity, 2 stop bits. 
200 bits/s means that each tone is 5 ms. 1180 Hz is a "1" and 980 is a "0". 

So, I need to get those 16 digits messages from tones back into 
numbers. So I need a way to listen the signals and analyse them and this
 is where I don't know how to do it. 

Maybe, I have to create a module into freeswitch and here also I need an help to do it. 

Thanks for your attention. 

8hector8
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From tonybecq at yahoo.fr  Mon Aug  6 18:33:44 2012
From: tonybecq at yahoo.fr (8hector8)
Date: Mon, 6 Aug 2012 07:33:44 -0700 (PDT)
Subject: [Freeswitch-users]  simulating a modem transaction...
Message-ID: <34260552.post@talk.nabble.com>


Hello everybody,

Here is my issue. I need to simulate a modem transaction in C. It's a V.21
modem in 200 bits/s. This modem dial a number and waits for a "request to
send" sent by the reciever. This is where I have to simulate a modem.
The "request to send" is composed of a 2100 Hz tone for 2,4 secondes and 32
alternations of two tones :
1650 Hz and 1850 Hz, 15 ms each.
In my idea, streamFile() function seams to be the good mean.

The real issue is that when those tones are sent, the transmeter will send a
"message" which is 16 digits in V.21 mode. 8 data bits, no parity, 2 stop
bits.
200 bits/s means that each tone is 5 ms. 1180 Hz is a "1" and 980 is a "0".

So, I need to get those 16 digits messages from tones back into numbers. To
do that, I think I need a way to listen the signals and analyse them and
this is where I don't know how to do it.

Maybe, I have to create a module into freeswitch and here also I need an
help to do it.

Thanks for your attention.

8hector8
-- 
View this message in context: http://old.nabble.com/simulating-a-modem-transaction...-tp34260552p34260552.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From jack at livecall.com  Mon Aug  6 19:04:53 2012
From: jack at livecall.com (Jack)
Date: Mon, 06 Aug 2012 08:04:53 -0700
Subject: [Freeswitch-users] RTMP Client not responding in Chrome
In-Reply-To: <501F4A4A.9040307@livecall.com>
References: <501F4A4A.9040307@livecall.com>
Message-ID: <501FDD15.1010405@livecall.com>

Here is the fix...    Go to chrome's Settings and disable the flash 
plugin and then go to adobe and download and re-install the flash plugin.


On 8/5/2012 9:38 PM, Jack wrote:
> Is anyone else experiencing problems with RTMP FLEX client in Chrome?
> Just today it started making chrome become unresponsive.  If I allow it
> to wait about 3 minutes, sometimes it finally starts working.
> I am experiencing this on my site as well as the conference.freeswitch
> .org/conf site.
> It is working fine in IE Safari, and Firefox.
> jack
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From bdfoster at endigotech.com  Mon Aug  6 19:54:01 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 6 Aug 2012 11:54:01 -0400
Subject: [Freeswitch-users] Fax (RX): Missing name/header variable in
	socket!?
In-Reply-To: <CALaOL+j3+pCk1x5=Wj6+sbDgyKeHW=eUZtCT4j-1zH+W_pJeBw@mail.gmail.com>
References: <CALaOL+hYAxkVTEOLBCpCZB5H6LbZaLH_hW-UiagvP_XYAKOQbQ@mail.gmail.com>
	<501BB72D.7070106@coppice.org>
	<CALaOL+h7-6pTtf0rVHD5X0wryawhyn7UZ5v87-mZ-jwXinE_gg@mail.gmail.com>
	<501BE1E6.1030101@coppice.org>
	<CALaOL+j3+pCk1x5=Wj6+sbDgyKeHW=eUZtCT4j-1zH+W_pJeBw@mail.gmail.com>
Message-ID: <CAKV-jFw1sanSOfF2HPnhm8f=oU42U4=-JHeH2PfFK6J0KwdTsg@mail.gmail.com>

it would more or less depending on the fax machine. i have a few fax
machines that do indeed send the remote station id as you describe, but i
have one that doesnt. from what im seeing, it wont matter what you use on
the receiving side the issue will still be there. good luck in your quest.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 6, 2012 5:27 AM, "Dennis" <odermann at googlemail.com> wrote:

> thanks, steve, then we will have to find something else.
>
> kind regards
> dennis
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing 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 yehavi.bourvine at gmail.com  Mon Aug  6 20:36:27 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 6 Aug 2012 19:36:27 +0300
Subject: [Freeswitch-users] Anyone using FS with Polycom and BLF (speed
	dials)?
Message-ID: <CAAtWKeFoSzy9ry6CrbirFj0jnLB+BkQWrF7LuHfQE2gDNGkOnQ@mail.gmail.com>

Hi,

  We have various Polycom phones which monitor other extensions. Most of
them use speed dials/BLF (dialog in SIP level) and some use the "buddies"
(presence at SIP level). The first method is preferable as the monitoring
phone can differentiate between the state of the line (mainly ringing as
opossed to busy).

In recent versions (after FS-2983) we fonud the following problem: When a
user puts a cal lon hold, dials another call, its state on the BLF
moitoring phone is stuck in BUSY. The buddies are ok.
 Has anyone else seen this problem? Or are we the only ones to use BLF on
Polycom...

I tried it on various 3.3 trains, and all behave the same.

BTW, on SNOM it is ok.

                         Thanks, __Yehavi:
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From ben at langfeld.co.uk  Mon Aug  6 22:52:42 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Mon, 6 Aug 2012 19:52:42 +0100
Subject: [Freeswitch-users] UniMRCP content type broken
Message-ID: <CAAyX+KH3bAPn4A__Ups2cZb8YsvmbfuW350efqgAA1wf0VUp1A@mail.gmail.com>

I'm testing FreeSWITCH with Voxeo PRISM's media server, via UniMRCP.
>From the FreeSWITCH side, it looks like everything works, but the
media server complains about an unknown content type, because contrary
to its logs, freeswitch appears to be sending application\sssml+xml
(too many 's').

Logs here: https://gist.github.com/477e53db58318125cc2c

Does anyone have this working, or know how to fix it?

Regards,
Ben Langfeld


From ben at langfeld.co.uk  Mon Aug  6 23:00:59 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Mon, 6 Aug 2012 20:00:59 +0100
Subject: [Freeswitch-users] UniMRCP content type broken
In-Reply-To: <CAAyX+KH3bAPn4A__Ups2cZb8YsvmbfuW350efqgAA1wf0VUp1A@mail.gmail.com>
References: <CAAyX+KH3bAPn4A__Ups2cZb8YsvmbfuW350efqgAA1wf0VUp1A@mail.gmail.com>
Message-ID: <CAAyX+KEt-+rOTUgcn=wabWnTmtP=Hf=2cQe3Re2oiokp-5cevA@mail.gmail.com>

JIRA here: http://jira.freeswitch.org/browse/FS-4500

Regards,
Ben Langfeld


On 6 August 2012 19:52, Ben Langfeld <ben at langfeld.co.uk> wrote:
> I'm testing FreeSWITCH with Voxeo PRISM's media server, via UniMRCP.
> From the FreeSWITCH side, it looks like everything works, but the
> media server complains about an unknown content type, because contrary
> to its logs, freeswitch appears to be sending application\sssml+xml
> (too many 's').
>
> Logs here: https://gist.github.com/477e53db58318125cc2c
>
> Does anyone have this working, or know how to fix it?
>
> Regards,
> Ben Langfeld


From Hector.Geraldino at ipsoft.com  Mon Aug  6 23:35:38 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Mon, 6 Aug 2012 15:35:38 -0400
Subject: [Freeswitch-users] UniMRCP content type broken
In-Reply-To: <CAAyX+KH3bAPn4A__Ups2cZb8YsvmbfuW350efqgAA1wf0VUp1A@mail.gmail.com>
References: <CAAyX+KH3bAPn4A__Ups2cZb8YsvmbfuW350efqgAA1wf0VUp1A@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD073E1FE8B0@NY1-EXMB-01.ip-soft.net>

What I can see is that the \s sequence is a character escape for the backslash. You can see the same pattern in your log:

<speak xmlns="http:\s\swww.w3.org\s2001\s10\ssynthesis" version="1.0" xml:lang="en-US">Hello<\sspeak>

Look all these \s sequences where the "/" should appear. I'm not sure FS is escaping these characters, I haven't experienced this issue when dealing with MRCP+Nuance Vocalizer for Network.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ben Langfeld
Sent: Monday, August 06, 2012 2:53 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] UniMRCP content type broken

I'm testing FreeSWITCH with Voxeo PRISM's media server, via UniMRCP.
>From the FreeSWITCH side, it looks like everything works, but the
media server complains about an unknown content type, because contrary to its logs, freeswitch appears to be sending application\sssml+xml (too many 's').

Logs here: https://gist.github.com/477e53db58318125cc2c

Does anyone have this working, or know how to fix it?

Regards,
Ben Langfeld

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


From ben at langfeld.co.uk  Tue Aug  7 00:31:41 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Mon, 6 Aug 2012 21:31:41 +0100
Subject: [Freeswitch-users] UniMRCP content type broken
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD073E1FE8B0@NY1-EXMB-01.ip-soft.net>
References: <CAAyX+KH3bAPn4A__Ups2cZb8YsvmbfuW350efqgAA1wf0VUp1A@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD073E1FE8B0@NY1-EXMB-01.ip-soft.net>
Message-ID: <CAAyX+KE4=jO5j3vbpqf09biCy5nYKmjZMmA2QUOpQNcA-gO8Xw@mail.gmail.com>

The SSML document itself is fine. It's the RTMP header that is the issue.

Regards,
Ben Langfeld


On 6 August 2012 20:35, Hector Geraldino <Hector.Geraldino at ipsoft.com> wrote:
> What I can see is that the \s sequence is a character escape for the backslash. You can see the same pattern in your log:
>
> <speak xmlns="http:\s\swww.w3.org\s2001\s10\ssynthesis" version="1.0" xml:lang="en-US">Hello<\sspeak>
>
> Look all these \s sequences where the "/" should appear. I'm not sure FS is escaping these characters, I haven't experienced this issue when dealing with MRCP+Nuance Vocalizer for Network.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ben Langfeld
> Sent: Monday, August 06, 2012 2:53 PM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] UniMRCP content type broken
>
> I'm testing FreeSWITCH with Voxeo PRISM's media server, via UniMRCP.
> >From the FreeSWITCH side, it looks like everything works, but the
> media server complains about an unknown content type, because contrary to its logs, freeswitch appears to be sending application\sssml+xml (too many 's').
>
> Logs here: https://gist.github.com/477e53db58318125cc2c
>
> Does anyone have this working, or know how to fix it?
>
> Regards,
> Ben Langfeld
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From vipkilla at gmail.com  Tue Aug  7 00:41:40 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 6 Aug 2012 16:41:40 -0400
Subject: [Freeswitch-users] Anyone using FS with Polycom and BLF (speed
	dials)?
In-Reply-To: <CAAtWKeFoSzy9ry6CrbirFj0jnLB+BkQWrF7LuHfQE2gDNGkOnQ@mail.gmail.com>
References: <CAAtWKeFoSzy9ry6CrbirFj0jnLB+BkQWrF7LuHfQE2gDNGkOnQ@mail.gmail.com>
Message-ID: <CAC-LwPOvOPAg1on=5qcbgvsE_-hYOxfTSBazDXrL380Ni4dQiw@mail.gmail.com>

I dont use the "buddies" but I do use BLF and SLA w/ Polycoms
I use the latest Polycom firmware (4.0X) and have not noticed any
issues as you've described.

On Mon, Aug 6, 2012 at 12:36 PM, Yehavi Bourvine
<yehavi.bourvine at gmail.com> wrote:
> Hi,
>
>   We have various Polycom phones which monitor other extensions. Most of
> them use speed dials/BLF (dialog in SIP level) and some use the "buddies"
> (presence at SIP level). The first method is preferable as the monitoring
> phone can differentiate between the state of the line (mainly ringing as
> opossed to busy).
>
> In recent versions (after FS-2983) we fonud the following problem: When a
> user puts a cal lon hold, dials another call, its state on the BLF moitoring
> phone is stuck in BUSY. The buddies are ok.
>  Has anyone else seen this problem? Or are we the only ones to use BLF on
> Polycom...
>
> I tried it on various 3.3 trains, and all behave the same.
>
> BTW, on SNOM it is ok.
>
>                          Thanks, __Yehavi:
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From casteven at gmail.com  Tue Aug  7 01:38:26 2012
From: casteven at gmail.com (Campbell Steven)
Date: Tue, 7 Aug 2012 09:38:26 +1200
Subject: [Freeswitch-users] Callcenter saying position in queue
In-Reply-To: <501FC899.5030601@gmail.com>
References: <501F86A3.4060904@gmail.com>
	<501FC899.5030601@gmail.com>
Message-ID: <CAPerZE9DqMYZhDZ2+BRg+4iJEifTVRRAETLFnGVfOwZ6Ko2E9w@mail.gmail.com>

Hi,

If you are using uuid_broadcast I suspect you have run into the same
issue as already open in this jira:

http://jira.freeswitch.org/browse/FS-4487

Please add your comments into the jira already open if it matches your
symptoms, the problem appears to be caused by a regression from a
change a few months back.

Campbell

On 7 August 2012 01:37, Vbvbrj <vbvbrj at gmail.com> wrote:
> So I managed to make a script in lua which will count callers position
> by uuid and say the position. But as soon as digit is playied back to
> caller, it is disconnected from callcenter application and hangs up. I
> want to play the digit together with the moh sound and caller will
> continue to stay in queue. Something which will transmit the digit
> playing in to the a-leg.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From trel.nadal at gmail.com  Tue Aug  7 00:09:45 2012
From: trel.nadal at gmail.com (Trel)
Date: Mon, 6 Aug 2012 13:09:45 -0700
Subject: [Freeswitch-users] google voice / XMPP questions!
Message-ID: <CAB3Kx7QYs6JQMTX8MqMas5URRqwb9Fcrar8iVcRMep9fDafvzg@mail.gmail.com>

I am using gvoice per the config and directions on the wiki:
http://wiki.freeswitch.org/wiki/Google_Voice
We're using a mixed bag of hardware SIP capable phones, linksys and
cisco mostly.

flow is: google voice <> ipcop firewall / NAT <> dmz'ed freeswitch
box(running current)  <> bunch of SIP phones

the freeswitch server is a dell poweredge, couple xeons at
3.something, four gig of ram, couple sata2 hard drives in raid1. load
stays under 14% basically all the time. iostat shows very low usage
for basically everything,

our pipe is comcast, 20/8.


we have, seemingly randomly, audio quality problems. most phone calls
are crisp and clear, and one in ten, (Sometimes in streaks of 3-4
calls) we'll have choppy sound both directions. "Hello my name is mike
and i'm calling for" sounds like: "llo ame is and alling"

this seems to be per-call, while one person in the office is having
poor call quality most others are fine at the same time.

i have enabled QoS, and prioritized traffic to and from the freeswitch
box. our pipe is not saturated during the problems.

i've played with VAD, send-silence, and rtp-timer. i've tried changing
the codec the phones use with no real effect, i've tried changing the
codec used in jingle but we're unable to dial out using anything but
PCMU.

Where should i look next?

Thanks in advance <3

Trel Nadal


From oseslija at gmail.com  Tue Aug  7 01:55:12 2012
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 6 Aug 2012 23:55:12 +0200
Subject: [Freeswitch-users] Early media: Sending 180 instead of 183?
In-Reply-To: <1FFF97C269757C458224B7C895F35F1513E262@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1513E262@cantor.std.visionutv.se>
Message-ID: <CAM21H-Mmk7fW2c3_jsQYK0dKffcRMOB9nu+Bz6rM5c6nYJGXCQ@mail.gmail.com>

180 should mean for the other side to provide ringing which is not
something you want I guess.
On 6 Aug 2012 13:37, "Peter Olsson" <peter.olsson at visionutveckling.se>
wrote:
>
> If you don't want to send 183, don't try to send early media. It's the
only way.
>
> In the dialplan you posted, remove the line: <action
application="playback" data="/tmp/test.wav"/>
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] F?r Dennis
> Skickat: den 6 augusti 2012 12:58
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Early media: Sending 180 instead of 183?
>
> we are doing a lot with ISDN-connections, while ISDN in germany seems to
be very different from the ISDN in the usa.
>
> our carriers asked us to only send a 180 instead of the 183 because, as i
wrote above, 183 messages are not mapped reliable to ISUP as written in the
specification ITU-T Rec. Q.1912.5.
> the a-leg sends an IAM or INVITE, but does not receive an answer - this
can differ from a-leg to a-leg.
>
> we know, that this problem is quite special, but we have the same
problems with different carriers.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 

>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:501fa15e32761376919998!
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 telefaks.de  Tue Aug  7 04:44:37 2012
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 07 Aug 2012 02:44:37 +0200
Subject: [Freeswitch-users] Problem with flex client connecting to mod_rtmp
Message-ID: <502064F5.2040702@telefaks.de>

Hello,

today I tried the flex client with the mod_rtmp implementation.
On a brandnew freeswitch I installed mod_rtmp and copied the flex
direrectory to a web server and loaded the web page.

However the flex client does not connect:
Here's my freeswitch log
2012-08-07 02:32:03.585671 [NOTICE] mod_rtmp.c:744 New RTMP session
[f35a192c-0d02-4320-89e9-775a3573ee25]
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:702 Sent handshake response
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:727 Done with handshake
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14
ts=0 stream_id=0x0] len=342
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE
for connect
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1
stream_id=0x0] len=4
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5
stream_id=0x0] len=4
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6
stream_id=0x0] len=5
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4
stream_id=0x0] len=6
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
stream_id=0x0] len=201
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
stream_id=0x0] len=61
2012-08-07 02:32:03.685670 [NOTICE] rtmp_sig.c:121 Sent connect reply
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5
ts=12247364 stream_id=0x0] len=4
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:945 Set window size: 131072 bytes

If I change the rtmp_url to 'rtmp://conference.freeswitch.org/phone',
flex does connect.

But in my freeswitch this fails. I have traced the network traffic and I
can see that there is information flow between freeswitch and the client
on connect request.

Somebody has an idea where to look further?

-- 
With kind regards
Peter 





From sdevoy at bizfocused.com  Tue Aug  7 06:34:20 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 6 Aug 2012 22:34:20 -0400
Subject: [Freeswitch-users] Smack a Hacker context?
Message-ID: <211601cd7445$26907820$73b16860$@bizfocused.com>

HI Everyone,

 

I didn't bother to obliterate my servers real identity from my last pastebin
post, shame on me.  Of course within 24 hours some a$$hole is trying the
default login ids for Freeswitch.  I am not quite a noob here, but this
would take me some time to do and I know some of you guys could crank this
out in minutes.  Of course you all probably have better ideas too.

 

I want to allow the users in the sample Freeswitch config to login - but to
special context.  In the "special" context, everything they dial plays a
recording like "you're an ass hangup and try again" or perhaps and ear
piercing LOUD tone.  Even better would be to dial them every few minutes
with the same recording or sound.  Best of all, if we can detect that they
are on a CISCO 5xx phone, I have a special config file to send them!

 

Anyone got better ideas?  I would love to hear them.

 

I suppose the adult choice would be to gather their external and internal ip
addresses and report them to their ISP, but that won't achieve much.  Maybe
we could build a blacklist and watch for them to connect to the Freeswitch
conf call.  Then we could all tell them how much we enjoy their efforts.

 

Sean

-------------- next part --------------
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From anthony.minessale at gmail.com  Tue Aug  7 07:04:11 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 6 Aug 2012 22:04:11 -0500
Subject: [Freeswitch-users] Smack a Hacker context?
In-Reply-To: <211601cd7445$26907820$73b16860$@bizfocused.com>
References: <211601cd7445$26907820$73b16860$@bizfocused.com>
Message-ID: <CAKbxfG85VCK8FL3eDSLU5TTYrRkOcV=qF4CD=KrZFWtER_xEEg@mail.gmail.com>

http://itslenny.com/

On Mon, Aug 6, 2012 at 9:34 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> HI Everyone,
>
>
>
> I didn?t bother to obliterate my servers real identity from my last pastebin
> post, shame on me.  Of course within 24 hours some a$$hole is trying the
> default login ids for Freeswitch.  I am not quite a noob here, but this
> would take me some time to do and I know some of you guys could crank this
> out in minutes.  Of course you all probably have better ideas too.
>
>
>
> I want to allow the users in the sample Freeswitch config to login ? but to
> special context.  In the ?special? context, everything they dial plays a
> recording like ?you?re an ass hangup and try again? or perhaps and ear
> piercing LOUD tone.  Even better would be to dial them every few minutes
> with the same recording or sound.  Best of all, if we can detect that they
> are on a CISCO 5xx phone, I have a special config file to send them!
>
>
>
> Anyone got better ideas?  I would love to hear them.
>
>
>
> I suppose the adult choice would be to gather their external and internal ip
> addresses and report them to their ISP, but that won?t achieve much.  Maybe
> we could build a blacklist and watch for them to connect to the Freeswitch
> conf call.  Then we could all tell them how much we enjoy their efforts.
>
>
>
> Sean
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 curriegrad2004 at gmail.com  Tue Aug  7 07:18:27 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 6 Aug 2012 20:18:27 -0700
Subject: [Freeswitch-users] Smack a Hacker context?
In-Reply-To: <CAKbxfG85VCK8FL3eDSLU5TTYrRkOcV=qF4CD=KrZFWtER_xEEg@mail.gmail.com>
References: <211601cd7445$26907820$73b16860$@bizfocused.com>
	<CAKbxfG85VCK8FL3eDSLU5TTYrRkOcV=qF4CD=KrZFWtER_xEEg@mail.gmail.com>
Message-ID: <CAMWiMy-wTDmKOoZUtW3_HgY-aC8LTQeAWd+=RN0VjuMn-RmN+A@mail.gmail.com>

Lenny's old.

You could just make up the credentials for the default users and set
the user-context to your "special" context. From there write that
context out and be a little imaginative :P

On Mon, Aug 6, 2012 at 8:04 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> http://itslenny.com/
>
> On Mon, Aug 6, 2012 at 9:34 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>> HI Everyone,
>>
>>
>>
>> I didn?t bother to obliterate my servers real identity from my last pastebin
>> post, shame on me.  Of course within 24 hours some a$$hole is trying the
>> default login ids for Freeswitch.  I am not quite a noob here, but this
>> would take me some time to do and I know some of you guys could crank this
>> out in minutes.  Of course you all probably have better ideas too.
>>
>>
>>
>> I want to allow the users in the sample Freeswitch config to login ? but to
>> special context.  In the ?special? context, everything they dial plays a
>> recording like ?you?re an ass hangup and try again? or perhaps and ear
>> piercing LOUD tone.  Even better would be to dial them every few minutes
>> with the same recording or sound.  Best of all, if we can detect that they
>> are on a CISCO 5xx phone, I have a special config file to send them!
>>
>>
>>
>> Anyone got better ideas?  I would love to hear them.
>>
>>
>>
>> I suppose the adult choice would be to gather their external and internal ip
>> addresses and report them to their ISP, but that won?t achieve much.  Maybe
>> we could build a blacklist and watch for them to connect to the Freeswitch
>> conf call.  Then we could all tell them how much we enjoy their efforts.
>>
>>
>>
>> Sean
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From chris at opencsta.org  Tue Aug  7 07:41:53 2012
From: chris at opencsta.org (Chris Mylonas)
Date: Tue, 7 Aug 2012 13:41:53 +1000
Subject: [Freeswitch-users] Smack a Hacker context?
In-Reply-To: <CAMWiMy-wTDmKOoZUtW3_HgY-aC8LTQeAWd+=RN0VjuMn-RmN+A@mail.gmail.com>
References: <211601cd7445$26907820$73b16860$@bizfocused.com>
	<CAKbxfG85VCK8FL3eDSLU5TTYrRkOcV=qF4CD=KrZFWtER_xEEg@mail.gmail.com>
	<CAMWiMy-wTDmKOoZUtW3_HgY-aC8LTQeAWd+=RN0VjuMn-RmN+A@mail.gmail.com>
Message-ID: <8BCDCD0B-6677-44F4-A70F-BDBCB1C208B4@opencsta.org>

hehe to lenny
just watch out for ear piercing noise.  you can can get sued for that
allow them to register and do nothing.  they will eventually scrub you off their list of targets.

IANAL & YMMV


On 07/08/2012, at 1:18 PM, curriegrad2004 wrote:

> Lenny's old.
> 
> You could just make up the credentials for the default users and set
> the user-context to your "special" context. From there write that
> context out and be a little imaginative :P
> 
> On Mon, Aug 6, 2012 at 8:04 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> http://itslenny.com/
>> 
>> On Mon, Aug 6, 2012 at 9:34 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>>> HI Everyone,
>>> 
>>> 
>>> 
>>> I didn?t bother to obliterate my servers real identity from my last pastebin
>>> post, shame on me.  Of course within 24 hours some a$$hole is trying the
>>> default login ids for Freeswitch.  I am not quite a noob here, but this
>>> would take me some time to do and I know some of you guys could crank this
>>> out in minutes.  Of course you all probably have better ideas too.
>>> 
>>> 
>>> 
>>> I want to allow the users in the sample Freeswitch config to login ? but to
>>> special context.  In the ?special? context, everything they dial plays a
>>> recording like ?you?re an ass hangup and try again? or perhaps and ear
>>> piercing LOUD tone.  Even better would be to dial them every few minutes
>>> with the same recording or sound.  Best of all, if we can detect that they
>>> are on a CISCO 5xx phone, I have a special config file to send them!
>>> 
>>> 
>>> 
>>> Anyone got better ideas?  I would love to hear them.
>>> 
>>> 
>>> 
>>> I suppose the adult choice would be to gather their external and internal ip
>>> addresses and report them to their ISP, but that won?t achieve much.  Maybe
>>> we could build a blacklist and watch for them to connect to the Freeswitch
>>> conf call.  Then we could all tell them how much we enjoy their efforts.
>>> 
>>> 
>>> 
>>> Sean
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> Join Us At ClueCon - Aug 7-9, 2012
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mitch.capper at gmail.com  Tue Aug  7 07:48:41 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 6 Aug 2012 20:48:41 -0700
Subject: [Freeswitch-users] Smack a Hacker context?
In-Reply-To: <8BCDCD0B-6677-44F4-A70F-BDBCB1C208B4@opencsta.org>
References: <211601cd7445$26907820$73b16860$@bizfocused.com>
	<CAKbxfG85VCK8FL3eDSLU5TTYrRkOcV=qF4CD=KrZFWtER_xEEg@mail.gmail.com>
	<CAMWiMy-wTDmKOoZUtW3_HgY-aC8LTQeAWd+=RN0VjuMn-RmN+A@mail.gmail.com>
	<8BCDCD0B-6677-44F4-A70F-BDBCB1C208B4@opencsta.org>
Message-ID: <CALn233pfWX5QeLzmS-otRTQC3ZVa-7DKaOJRvbJie2RMvFXQxQ@mail.gmail.com>

Just have the dialplan send the users to the one tonestream you want,
also next time take a look at
http://wiki.freeswitch.org/wiki/Fs_logger.pl not only can it autoscrub
most of the data for you but you can specify an additional dictionary
of stuff for it to scrub prior to PBing something.

~mitch

On Mon, Aug 6, 2012 at 8:41 PM, Chris Mylonas <chris at opencsta.org> wrote:
> hehe to lenny
> just watch out for ear piercing noise.  you can can get sued for that
> allow them to register and do nothing.  they will eventually scrub you off their list of targets.
>
> IANAL & YMMV
>
>
> On 07/08/2012, at 1:18 PM, curriegrad2004 wrote:
>
>> Lenny's old.
>>
>> You could just make up the credentials for the default users and set
>> the user-context to your "special" context. From there write that
>> context out and be a little imaginative :P
>>
>> On Mon, Aug 6, 2012 at 8:04 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> http://itslenny.com/
>>>
>>> On Mon, Aug 6, 2012 at 9:34 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>>>> HI Everyone,
>>>>
>>>>
>>>>
>>>> I didn?t bother to obliterate my servers real identity from my last pastebin
>>>> post, shame on me.  Of course within 24 hours some a$$hole is trying the
>>>> default login ids for Freeswitch.  I am not quite a noob here, but this
>>>> would take me some time to do and I know some of you guys could crank this
>>>> out in minutes.  Of course you all probably have better ideas too.
>>>>
>>>>
>>>>
>>>> I want to allow the users in the sample Freeswitch config to login ? but to
>>>> special context.  In the ?special? context, everything they dial plays a
>>>> recording like ?you?re an ass hangup and try again? or perhaps and ear
>>>> piercing LOUD tone.  Even better would be to dial them every few minutes
>>>> with the same recording or sound.  Best of all, if we can detect that they
>>>> are on a CISCO 5xx phone, I have a special config file to send them!
>>>>
>>>>
>>>>
>>>> Anyone got better ideas?  I would love to hear them.
>>>>
>>>>
>>>>
>>>> I suppose the adult choice would be to gather their external and internal ip
>>>> addresses and report them to their ISP, but that won?t achieve much.  Maybe
>>>> we could build a blacklist and watch for them to connect to the Freeswitch
>>>> conf call.  Then we could all tell them how much we enjoy their efforts.
>>>>
>>>>
>>>>
>>>> Sean
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From jack at livecall.com  Tue Aug  7 08:26:41 2012
From: jack at livecall.com (Jack)
Date: Mon, 06 Aug 2012 21:26:41 -0700
Subject: [Freeswitch-users] Problem with flex client connecting to
	mod_rtmp
In-Reply-To: <502064F5.2040702@telefaks.de>
References: <502064F5.2040702@telefaks.de>
Message-ID: <50209901.10107@livecall.com>

Hi Peter,
Make sure you use a fully qualified username not just the extension 
1001 at xxx.xxx.xxx.xxx
the XXX would be the IP of your FreeSwitch Server.

did you configure your rtmp.conf.xml and have a matching context in your 
dial plan?

In the web page make sure these two vars are set to YOUR Freeswitch 
server ip:
var rtmpIPURL = "rtmp://xxx.xxx.xxx.xxx/phone";
  var rtmpIP="xxx.xxx.xxx.xxx";

jack

On 8/6/2012 5:44 PM, Peter Steinbach wrote:
> Hello,
>
> today I tried the flex client with the mod_rtmp implementation.
> On a brandnew freeswitch I installed mod_rtmp and copied the flex
> direrectory to a web server and loaded the web page.
>
> However the flex client does not connect:
> Here's my freeswitch log
> 2012-08-07 02:32:03.585671 [NOTICE] mod_rtmp.c:744 New RTMP session
> [f35a192c-0d02-4320-89e9-775a3573ee25]
> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:702 Sent handshake response
> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:727 Done with handshake
> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14
> ts=0 stream_id=0x0] len=342
> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE
> for connect
> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1
> stream_id=0x0] len=4
> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5
> stream_id=0x0] len=4
> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6
> stream_id=0x0] len=5
> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4
> stream_id=0x0] len=6
> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
> stream_id=0x0] len=201
> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
> stream_id=0x0] len=61
> 2012-08-07 02:32:03.685670 [NOTICE] rtmp_sig.c:121 Sent connect reply
> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5
> ts=12247364 stream_id=0x0] len=4
> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
>
> If I change the rtmp_url to 'rtmp://conference.freeswitch.org/phone',
> flex does connect.
>
> But in my freeswitch this fails. I have traced the network traffic and I
> can see that there is information flow between freeswitch and the client
> on connect request.
>
> Somebody has an idea where to look further?
>



From vbvbrj at gmail.com  Tue Aug  7 09:27:28 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Tue, 07 Aug 2012 08:27:28 +0300
Subject: [Freeswitch-users] Callcenter saying position in queue
In-Reply-To: <CAPerZE9DqMYZhDZ2+BRg+4iJEifTVRRAETLFnGVfOwZ6Ko2E9w@mail.gmail.com>
References: <501F86A3.4060904@gmail.com> <501FC899.5030601@gmail.com>
	<CAPerZE9DqMYZhDZ2+BRg+4iJEifTVRRAETLFnGVfOwZ6Ko2E9w@mail.gmail.com>
Message-ID: <5020A740.5030007@gmail.com>

On 07.08.2012 00:38, Campbell Steven wrote:
> Hi,
>
> If you are using uuid_broadcast I suspect you have run into the same
> issue as already open in this jira:
>
> http://jira.freeswitch.org/browse/FS-4487
>
> Please add your comments into the jira already open if it matches your
> symptoms, the problem appears to be caused by a regression from a
> change a few months back.
>
> Campbell


I added comment. Hope it is the same and indeed is a problem with 
regression, as as wiki says about uuid_broadcast that it must just 
displace the sound and insert another. A better will be if it will be muxed.



From peter.olsson at visionutveckling.se  Tue Aug  7 10:02:12 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 7 Aug 2012 06:02:12 +0000
Subject: [Freeswitch-users] Callcenter saying position in queue
In-Reply-To: <5020A740.5030007@gmail.com>
References: <501F86A3.4060904@gmail.com> <501FC899.5030601@gmail.com>
	<CAPerZE9DqMYZhDZ2+BRg+4iJEifTVRRAETLFnGVfOwZ6Ko2E9w@mail.gmail.com>,
	<5020A740.5030007@gmail.com>
Message-ID: <ABD2F67B-8E2F-45E7-A9AD-91E6FFE58428@visionutveckling.se>

Did you notice this was resolved already? Did you try latest yet?

/Peter

7 aug 2012 kl. 07:33 skrev "Vbvbrj" <vbvbrj at gmail.com>:

> On 07.08.2012 00:38, Campbell Steven wrote:
>> Hi,
>> 
>> If you are using uuid_broadcast I suspect you have run into the same
>> issue as already open in this jira:
>> 
>> http://jira.freeswitch.org/browse/FS-4487
>> 
>> Please add your comments into the jira already open if it matches your
>> symptoms, the problem appears to be caused by a regression from a
>> change a few months back.
>> 
>> Campbell
> 
> 
> I added comment. Hope it is the same and indeed is a problem with 
> regression, as as wiki says about uuid_broadcast that it must just 
> displace the sound and insert another. A better will be if it will be muxed.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:5020a54632761721255575!
> 


From vbvbrj at gmail.com  Tue Aug  7 10:29:11 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Tue, 07 Aug 2012 09:29:11 +0300
Subject: [Freeswitch-users] Callcenter saying position in queue
In-Reply-To: <ABD2F67B-8E2F-45E7-A9AD-91E6FFE58428@visionutveckling.se>
References: <501F86A3.4060904@gmail.com> <501FC899.5030601@gmail.com>
	<CAPerZE9DqMYZhDZ2+BRg+4iJEifTVRRAETLFnGVfOwZ6Ko2E9w@mail.gmail.com>,
	<5020A740.5030007@gmail.com>
	<ABD2F67B-8E2F-45E7-A9AD-91E6FFE58428@visionutveckling.se>
Message-ID: <5020B5B7.4030105@gmail.com>

On 07.08.2012 09:02, Peter Olsson wrote:
> Did you notice this was resolved already? Did you try latest yet?
>
> /Peter

Yes. And I am trying it now. Building and installing the latest. After a 
test I will report back.



From vbvbrj at gmail.com  Tue Aug  7 10:48:58 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Tue, 07 Aug 2012 09:48:58 +0300
Subject: [Freeswitch-users] Callcenter saying position in queue
In-Reply-To: <ABD2F67B-8E2F-45E7-A9AD-91E6FFE58428@visionutveckling.se>
References: <501F86A3.4060904@gmail.com> <501FC899.5030601@gmail.com>
	<CAPerZE9DqMYZhDZ2+BRg+4iJEifTVRRAETLFnGVfOwZ6Ko2E9w@mail.gmail.com>,
	<5020A740.5030007@gmail.com>
	<ABD2F67B-8E2F-45E7-A9AD-91E6FFE58428@visionutveckling.se>
Message-ID: <5020BA5A.5030804@gmail.com>

On 07.08.2012 09:02, Peter Olsson wrote:
> Did you notice this was resolved already? Did you try latest yet?
>
> /Peter

I can confirm that the problem persists.



From khuenm at vega.com.vn  Tue Aug  7 06:24:50 2012
From: khuenm at vega.com.vn (Khue Nguyen Minh)
Date: Tue, 7 Aug 2012 09:24:50 +0700
Subject: [Freeswitch-users] Make outbound call
Message-ID: <CAO-yT6hrFqT_hKaqO41GWppsVqHbcSBwkNbMUg1R1Ue03JLAZw@mail.gmail.com>

Hi all,

I am using event socket of FreeSWITCH in "outbound" mode. From my
application, I want request FreeSWITCH make outbound call to a SIP client.
How I can do it, please guide me?

Best regards,
Khue Nguyen.
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From lists at telefaks.de  Tue Aug  7 12:11:37 2012
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 07 Aug 2012 10:11:37 +0200
Subject: [Freeswitch-users] Problem with flex client connecting to
	mod_rtmp
In-Reply-To: <50209901.10107@livecall.com>
References: <502064F5.2040702@telefaks.de> <50209901.10107@livecall.com>
Message-ID: <5020CDB9.3060102@telefaks.de>

Hello Jack,

here are some answers to your your questions:

> Make sure you use a fully qualified username not just the extension
>1001 at xxx.xxx.xxx.xxx >the XXX would be the IP of your FreeSwitch Server.
I am still in the "connect" state, I am not logging in yet. The webpage
tells me "Connecting... " and waits forever

> did you configure your rtmp.conf.xml and have a matching context in
your dial plan?
I am still in the "connect" state

In the web page make sure these two vars are set to YOUR Freeswitch
server ip: var rtmpIPURL =
> "rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx";
I did not have a rtmpIP var, so I added it to the code, but no change.

In fact as I can see while grepping the network traffic, the Flex app
and Freeswicth are negociating their protocol and freeswitch logs a
"Sent connect reply", but this is very diferent from what is sent from
the conference.freeswitch.org server.

Best regards
Peter


On 08/07/12 06:26, Jack wrote:
> Hi Peter,
> Make sure you use a fully qualified username not just the extension 
> 1001 at xxx.xxx.xxx.xxx
> the XXX would be the IP of your FreeSwitch Server.
>
> did you configure your rtmp.conf.xml and have a matching context in your 
> dial plan?
>
> In the web page make sure these two vars are set to YOUR Freeswitch 
> server ip:
> var rtmpIPURL = "rtmp://xxx.xxx.xxx.xxx/phone";
>   var rtmpIP="xxx.xxx.xxx.xxx";
>
> jack
>
> On 8/6/2012 5:44 PM, Peter Steinbach wrote:
>> Hello,
>>
>> today I tried the flex client with the mod_rtmp implementation.
>> On a brandnew freeswitch I installed mod_rtmp and copied the flex
>> direrectory to a web server and loaded the web page.
>>
>> However the flex client does not connect:
>> Here's my freeswitch log
>> 2012-08-07 02:32:03.585671 [NOTICE] mod_rtmp.c:744 New RTMP session
>> [f35a192c-0d02-4320-89e9-775a3573ee25]
>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:702 Sent handshake response
>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:727 Done with handshake
>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14
>> ts=0 stream_id=0x0] len=342
>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE
>> for connect
>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1
>> stream_id=0x0] len=4
>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5
>> stream_id=0x0] len=4
>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6
>> stream_id=0x0] len=5
>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4
>> stream_id=0x0] len=6
>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
>> stream_id=0x0] len=201
>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
>> stream_id=0x0] len=61
>> 2012-08-07 02:32:03.685670 [NOTICE] rtmp_sig.c:121 Sent connect reply
>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5
>> ts=12247364 stream_id=0x0] len=4
>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
>>
>> If I change the rtmp_url to 'rtmp://conference.freeswitch.org/phone',
>> flex does connect.
>>
>> But in my freeswitch this fails. I have traced the network traffic and I
>> can see that there is information flow between freeswitch and the client
>> on connect request.
>>
>> Somebody has an idea where to look further?
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From m.prevail at gmail.com  Tue Aug  7 12:16:18 2012
From: m.prevail at gmail.com (Prevail Magid)
Date: Tue, 7 Aug 2012 11:16:18 +0300
Subject: [Freeswitch-users] Make outbound call
In-Reply-To: <CAO-yT6hrFqT_hKaqO41GWppsVqHbcSBwkNbMUg1R1Ue03JLAZw@mail.gmail.com>
References: <CAO-yT6hrFqT_hKaqO41GWppsVqHbcSBwkNbMUg1R1Ue03JLAZw@mail.gmail.com>
Message-ID: <CANJXrA9nxtDTLUuOZRi270Dmy=-s1FowXBXfAtjgw73t7eQ62Q@mail.gmail.com>

You can do it by several way.

For example:

sendAsyncCommand( channel, "bgapi originate user/1001 777");
http://wiki.freeswitch.org/wiki/Mod_commands#originate


2012/8/7 Khue Nguyen Minh <khuenm at vega.com.vn>

> Hi all,
>
> I am using event socket of FreeSWITCH in "outbound" mode. From my
> application, I want request FreeSWITCH make outbound call to a SIP client.
> How I can do it, please guide me?
>
> Best regards,
> Khue Nguyen.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From miha at softnet.si  Tue Aug  7 12:36:59 2012
From: miha at softnet.si (Miha)
Date: Tue, 07 Aug 2012 10:36:59 +0200
Subject: [Freeswitch-users] upadte FS (make current)
Message-ID: <5020D3AB.7040401@softnet.si>

Hi,

I am running on production server FS (runs ok). I would like to upgrade 
it to latest git.
Can I expect any problems? I not using any special module (just 
mod_rad_auth, for cdr mod_xml_cdr).

Is it better to first test everything on test server and than make upgrade?

My FS is now up for 13 days. I am looking at memory for a few days and 
noticing that free memory is shrinking (slowly). Was there any problem 
with memory leak? :

top - 09:46:46 up 13 days, 12:32,  3 users,  load average: 0.00, 0.04, 0.05
Tasks: 159 total,   1 running, 158 sleeping,   0 stopped,   0 zombie
Cpu(s):  0.6%us,  0.4%sy,  0.0%ni, 99.0%id,  0.0%wa,  0.0%hi, 0.0%si,  
0.0%st
Mem:   4046636k total,  3356340k used,   690296k free,   357296k buffers
Swap:  6094840k total,        0k used,  6094840k free,  2629280k cached

My version: FreeSWITCH Version 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5 
(1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul 2012 15:18:27 Z)

Thanks!

Miha


From peter.olsson at visionutveckling.se  Tue Aug  7 12:54:06 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 7 Aug 2012 08:54:06 +0000
Subject: [Freeswitch-users] upadte FS (make current)
Message-ID: <1FFF97C269757C458224B7C895F35F1513F43E@cantor.std.visionutv.se>

Usually it's no problem. However, I always recommend to have a test server to run basic tests before going into production. There have been a couple of memory leaks fixed, but I think that was before your current version - but I don't remember all the commits from my head.. :)

I would also suggest to read through the git commit log, to get an idea of what has been done.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
Skickat: den 7 augusti 2012 10:37
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] upadte FS (make current)

Hi,

I am running on production server FS (runs ok). I would like to upgrade it to latest git.
Can I expect any problems? I not using any special module (just mod_rad_auth, for cdr mod_xml_cdr).

Is it better to first test everything on test server and than make upgrade?

My FS is now up for 13 days. I am looking at memory for a few days and noticing that free memory is shrinking (slowly). Was there any problem with memory leak? :

top - 09:46:46 up 13 days, 12:32,  3 users,  load average: 0.00, 0.04, 0.05
Tasks: 159 total,   1 running, 158 sleeping,   0 stopped,   0 zombie
Cpu(s):  0.6%us,  0.4%sy,  0.0%ni, 99.0%id,  0.0%wa,  0.0%hi, 0.0%si, 0.0%st
Mem:   4046636k total,  3356340k used,   690296k free,   357296k buffers
Swap:  6094840k total,        0k used,  6094840k free,  2629280k cached

My version: FreeSWITCH Version 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5
(1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul 2012 15:18:27 Z)

Thanks!

Miha

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at 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:5020d1a332763868583022!



From miha at softnet.si  Tue Aug  7 13:14:17 2012
From: miha at softnet.si (Miha)
Date: Tue, 07 Aug 2012 11:14:17 +0200
Subject: [Freeswitch-users] upadte FS (make current)
In-Reply-To: <1FFF97C269757C458224B7C895F35F1513F43E@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1513F43E@cantor.std.visionutv.se>
Message-ID: <5020DC69.5090909@softnet.si>

On 8/7/2012 10:54 AM, Peter Olsson wrote:
> Usually it's no problem. However, I always recommend to have a test server to run basic tests before going into production. There have been a couple of memory leaks fixed, but I think that was before your current version - but I don't remember all the commits from my head.. :)
>
> I would also suggest to read through the git commit log, to get an idea of what has been done.
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
> Skickat: den 7 augusti 2012 10:37
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] upadte FS (make current)
>
> Hi,
>
> I am running on production server FS (runs ok). I would like to upgrade it to latest git.
> Can I expect any problems? I not using any special module (just mod_rad_auth, for cdr mod_xml_cdr).
>
> Is it better to first test everything on test server and than make upgrade?
>
> My FS is now up for 13 days. I am looking at memory for a few days and noticing that free memory is shrinking (slowly). Was there any problem with memory leak? :
>
> top - 09:46:46 up 13 days, 12:32,  3 users,  load average: 0.00, 0.04, 0.05
> Tasks: 159 total,   1 running, 158 sleeping,   0 stopped,   0 zombie
> Cpu(s):  0.6%us,  0.4%sy,  0.0%ni, 99.0%id,  0.0%wa,  0.0%hi, 0.0%si, 0.0%st
> Mem:   4046636k total,  3356340k used,   690296k free,   357296k buffers
> Swap:  6094840k total,        0k used,  6094840k free,  2629280k cached
>
> My version: FreeSWITCH Version 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5
> (1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul 2012 15:18:27 Z)
>
> Thanks!
>
> Miha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:5020d1a332763868583022!
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
@Peter thanks for your quick answer:)

I readed jira postes and noticed that this was befor my git version (as 
you said).
If there was leak problem (I do not know if it is, just noticing that 
free memory is droping), would FS be stable for 13 days or would this 
bug be present sooner (memory leak)?

regards,
Miha



From asilva at wirelessmundi.com  Tue Aug  7 13:35:50 2012
From: asilva at wirelessmundi.com (Antonio)
Date: Tue, 07 Aug 2012 11:35:50 +0200
Subject: [Freeswitch-users] Problem with flex client connecting to
 mod_rtmp
In-Reply-To: <5020CDB9.3060102@telefaks.de>
References: <502064F5.2040702@telefaks.de> <50209901.10107@livecall.com>
	<5020CDB9.3060102@telefaks.de>
Message-ID: <1344332150.693.7.camel@marces.madrid.commsmundi.com>

Hi,

have you try putting:

var flashvars = { 
rtmp_url: 'rtmp://xxx.xxx.xxx.xxx' 
};
var params  = {
allowScriptAccess: 'always'
};

Also, you should remove the "phone" part from the url.


Regards,
Antonio


On Tue, 2012-08-07 at 10:11 +0200, Peter Steinbach wrote: 

> Hello Jack,
> 
> here are some answers to your your questions:
> 
> > Make sure you use a fully qualified username not just the extension
> >1001 at xxx.xxx.xxx.xxx >the XXX would be the IP of your FreeSwitch Server.
> I am still in the "connect" state, I am not logging in yet. The webpage
> tells me "Connecting... " and waits forever
> 
> > did you configure your rtmp.conf.xml and have a matching context in
> your dial plan?
> I am still in the "connect" state
> 
> In the web page make sure these two vars are set to YOUR Freeswitch
> server ip: var rtmpIPURL =
> > "rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx";
> I did not have a rtmpIP var, so I added it to the code, but no change.
> 
> In fact as I can see while grepping the network traffic, the Flex app
> and Freeswicth are negociating their protocol and freeswitch logs a
> "Sent connect reply", but this is very diferent from what is sent from
> the conference.freeswitch.org server.
> 
> Best regards
> Peter
> 
> 
> On 08/07/12 06:26, Jack wrote:
> > Hi Peter,
> > Make sure you use a fully qualified username not just the extension 
> > 1001 at xxx.xxx.xxx.xxx
> > the XXX would be the IP of your FreeSwitch Server.
> >
> > did you configure your rtmp.conf.xml and have a matching context in your 
> > dial plan?
> >
> > In the web page make sure these two vars are set to YOUR Freeswitch 
> > server ip:
> > var rtmpIPURL = "rtmp://xxx.xxx.xxx.xxx/phone";
> >   var rtmpIP="xxx.xxx.xxx.xxx";
> >
> > jack
> >
> > On 8/6/2012 5:44 PM, Peter Steinbach wrote:
> >> Hello,
> >>
> >> today I tried the flex client with the mod_rtmp implementation.
> >> On a brandnew freeswitch I installed mod_rtmp and copied the flex
> >> direrectory to a web server and loaded the web page.
> >>
> >> However the flex client does not connect:
> >> Here's my freeswitch log
> >> 2012-08-07 02:32:03.585671 [NOTICE] mod_rtmp.c:744 New RTMP session
> >> [f35a192c-0d02-4320-89e9-775a3573ee25]
> >> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:702 Sent handshake response
> >> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:727 Done with handshake
> >> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14
> >> ts=0 stream_id=0x0] len=342
> >> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE
> >> for connect
> >> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1
> >> stream_id=0x0] len=4
> >> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5
> >> stream_id=0x0] len=4
> >> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6
> >> stream_id=0x0] len=5
> >> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4
> >> stream_id=0x0] len=6
> >> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
> >> stream_id=0x0] len=201
> >> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
> >> stream_id=0x0] len=61
> >> 2012-08-07 02:32:03.685670 [NOTICE] rtmp_sig.c:121 Sent connect reply
> >> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5
> >> ts=12247364 stream_id=0x0] len=4
> >> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
> >>
> >> If I change the rtmp_url to 'rtmp://conference.freeswitch.org/phone',
> >> flex does connect.
> >>
> >> But in my freeswitch this fails. I have traced the network traffic and I
> >> can see that there is information flow between freeswitch and the client
> >> on connect request.
> >>
> >> Somebody has an idea where to look further?
> >>
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 

Un cordial saludo / Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com
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From peter.olsson at visionutveckling.se  Tue Aug  7 13:48:20 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 7 Aug 2012 09:48:20 +0000
Subject: [Freeswitch-users] upadte FS (make current)
Message-ID: <1FFF97C269757C458224B7C895F35F1513F759@cantor.std.visionutv.se>

It's always hard to say what is a memory leak and what's not. FS uses memory pools internally, which sometime might make it look like it leaks memory, but in the end FS will reuse that memory. You can also run "fsctl reclaim_mem" in the console or fs_cli, and see if that makes the memory usage drop.

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
Skickat: den 7 augusti 2012 11:14
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] upadte FS (make current)

On 8/7/2012 10:54 AM, Peter Olsson wrote:
> Usually it's no problem. However, I always recommend to have a test 
> server to run basic tests before going into production. There have 
> been a couple of memory leaks fixed, but I think that was before your 
> current version - but I don't remember all the commits from my head.. 
> :)
>
> I would also suggest to read through the git commit log, to get an idea of what has been done.
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
> Skickat: den 7 augusti 2012 10:37
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] upadte FS (make current)
>
> Hi,
>
> I am running on production server FS (runs ok). I would like to upgrade it to latest git.
> Can I expect any problems? I not using any special module (just mod_rad_auth, for cdr mod_xml_cdr).
>
> Is it better to first test everything on test server and than make upgrade?
>
> My FS is now up for 13 days. I am looking at memory for a few days and noticing that free memory is shrinking (slowly). Was there any problem with memory leak? :
>
> top - 09:46:46 up 13 days, 12:32,  3 users,  load average: 0.00, 0.04, 0.05
> Tasks: 159 total,   1 running, 158 sleeping,   0 stopped,   0 zombie
> Cpu(s):  0.6%us,  0.4%sy,  0.0%ni, 99.0%id,  0.0%wa,  0.0%hi, 0.0%si, 0.0%st
> Mem:   4046636k total,  3356340k used,   690296k free,   357296k buffers
> Swap:  6094840k total,        0k used,  6094840k free,  2629280k cached
>
> My version: FreeSWITCH Version 
> 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5
> (1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul 2012 15:18:27 Z)
>
> Thanks!
>
> Miha
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
> 
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
@Peter thanks for your quick answer:)

I readed jira postes and noticed that this was befor my git version (as you said).
If there was leak problem (I do not know if it is, just noticing that free memory is droping), would FS be stable for 13 days or would this bug be present sooner (memory leak)?

regards,
Miha


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at 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:5020daa032762709953773!



From Adam.Lappe at qsc.de  Tue Aug  7 13:48:56 2012
From: Adam.Lappe at qsc.de (Lappe, Adam)
Date: Tue, 7 Aug 2012 11:48:56 +0200
Subject: [Freeswitch-users] sendevent to Gateway
In-Reply-To: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EBF@CGNMSG01.oc.qsc.de>
References: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EBF@CGNMSG01.oc.qsc.de>
Message-ID: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EC1@CGNMSG01.oc.qsc.de>

I am still looking for a solution.
Is there no one who has an idea?

Thanks,
Adam

________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Lappe, Adam
Gesendet: Freitag, 3. August 2012 11:53
An: 'FreeSWITCH-users at lists.freeswitch.org'
Betreff: [Freeswitch-users] sendevent to Gateway

Hi all,

i am trying to make the freeswitch send a event (or SIP Message) to my gateway.
I can only find examples how to send events to registered endpoint but not to my gateway.

Shouldn?t it be possible to do this via the event socket?

Thanks in advance and best regards,
Adam
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From david.villasmil.work at gmail.com  Tue Aug  7 13:59:02 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 7 Aug 2012 11:59:02 +0200
Subject: [Freeswitch-users] sendevent to Gateway
In-Reply-To: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EC1@CGNMSG01.oc.qsc.de>
References: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EBF@CGNMSG01.oc.qsc.de>
	<F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EC1@CGNMSG01.oc.qsc.de>
Message-ID: <CAFGRPVro3EE2ug57k4bkmZHPGbF1EcquQeN_vtQ08dKs-nAxMQ@mail.gmail.com>

Hello,

What EXCATLY do you want to send?

David

On Tue, Aug 7, 2012 at 11:48 AM, Lappe, Adam <Adam.Lappe at qsc.de> wrote:

>  I am still looking for a solution.****
>
> Is there no one who has an idea?****
>
> ** **
>
> Thanks,****
>
> Adam****
>
> ** **
>  ------------------------------
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Lappe,
> Adam
> *Gesendet:* Freitag, 3. August 2012 11:53
> *An:* 'FreeSWITCH-users at lists.freeswitch.org'
> *Betreff:* [Freeswitch-users] sendevent to Gateway****
>
> ** **
>
> Hi all,****
>
> ** **
>
> i am trying to make the freeswitch send a event (or SIP Message) to my
> gateway.****
>
> I can only find examples how to send events to registered endpoint but not
> to my gateway.****
>
> ** **
>
> Shouldn?t it be possible to do this via the event socket?****
>
> ** **
>
> Thanks in advance and best regards,****
>
> Adam****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From miha at softnet.si  Tue Aug  7 14:21:30 2012
From: miha at softnet.si (Miha)
Date: Tue, 07 Aug 2012 12:21:30 +0200
Subject: [Freeswitch-users] upadte FS (make current)
In-Reply-To: <1FFF97C269757C458224B7C895F35F1513F759@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1513F759@cantor.std.visionutv.se>
Message-ID: <5020EC2A.2080302@softnet.si>

Hi Peter,

fsctl reclaim_mem this will not harm any call and etc?

So if freeswitch does not leak memory what should I see in top (memory will not drop?)?

Thanks!
Miha



On 8/7/2012 11:48 AM, Peter Olsson wrote:
> It's always hard to say what is a memory leak and what's not. FS uses memory pools internally, which sometime might make it look like it leaks memory, but in the end FS will reuse that memory. You can also run "fsctl reclaim_mem" in the console or fs_cli, and see if that makes the memory usage drop.
>
> /Peter
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
> Skickat: den 7 augusti 2012 11:14
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] upadte FS (make current)
>
> On 8/7/2012 10:54 AM, Peter Olsson wrote:
>> Usually it's no problem. However, I always recommend to have a test
>> server to run basic tests before going into production. There have
>> been a couple of memory leaks fixed, but I think that was before your
>> current version - but I don't remember all the commits from my head..
>> :)
>>
>> I would also suggest to read through the git commit log, to get an idea of what has been done.
>>
>> /Peter
>>
>>
>> -----Ursprungligt meddelande-----
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
>> Skickat: den 7 augusti 2012 10:37
>> Till: FreeSWITCH Users Help
>> ?mne: [Freeswitch-users] upadte FS (make current)
>>
>> Hi,
>>
>> I am running on production server FS (runs ok). I would like to upgrade it to latest git.
>> Can I expect any problems? I not using any special module (just mod_rad_auth, for cdr mod_xml_cdr).
>>
>> Is it better to first test everything on test server and than make upgrade?
>>
>> My FS is now up for 13 days. I am looking at memory for a few days and noticing that free memory is shrinking (slowly). Was there any problem with memory leak? :
>>
>> top - 09:46:46 up 13 days, 12:32,  3 users,  load average: 0.00, 0.04, 0.05
>> Tasks: 159 total,   1 running, 158 sleeping,   0 stopped,   0 zombie
>> Cpu(s):  0.6%us,  0.4%sy,  0.0%ni, 99.0%id,  0.0%wa,  0.0%hi, 0.0%si, 0.0%st
>> Mem:   4046636k total,  3356340k used,   690296k free,   357296k buffers
>> Swap:  6094840k total,        0k used,  6094840k free,  2629280k cached
>>
>> My version: FreeSWITCH Version
>> 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5
>> (1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul 2012 15:18:27 Z)
>>
>> Thanks!
>>
>> Miha
>>
>> ______________________________________________________________________
>> ___ Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
>>
>>
>>
>> ______________________________________________________________________
>> ___ Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
> @Peter thanks for your quick answer:)
>
> I readed jira postes and noticed that this was befor my git version (as you said).
> If there was leak problem (I do not know if it is, just noticing that free memory is droping), would FS be stable for 13 days or would this bug be present sooner (memory leak)?
>
> regards,
> Miha
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:5020daa032762709953773!
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From lists at telefaks.de  Tue Aug  7 14:26:31 2012
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 07 Aug 2012 12:26:31 +0200
Subject: [Freeswitch-users] Problem with flex client connecting to
	mod_rtmp
In-Reply-To: <1344332150.693.7.camel@marces.madrid.commsmundi.com>
References: <502064F5.2040702@telefaks.de> <50209901.10107@livecall.com>
	<5020CDB9.3060102@telefaks.de>
	<1344332150.693.7.camel@marces.madrid.commsmundi.com>
Message-ID: <5020ED57.1040505@telefaks.de>

Hello Antonio,

I added this:
> var params  = {
> allowScriptAccess: 'always'
> };

and removed the /phone from the URL. This did not change anything. Log
is still like this:

2012-08-07 11:56:48.105667 [NOTICE] mod_rtmp.c:744 New RTMP session
[b31edb8a-50d3-4100-9b7f-ab49b6ba06b5]
2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:702 Sent handshake response
2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:727 Done with handshake
2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14
ts=0 stream_id=0x0] len=331
2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE
for connect
2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1
stream_id=0x0] len=4
2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5
stream_id=0x0] len=4
2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6
stream_id=0x0] len=5
2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4
stream_id=0x0] len=6
2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
stream_id=0x0] len=201
2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
stream_id=0x0] len=61
2012-08-07 11:56:48.205670 [NOTICE] rtmp_sig.c:121 Sent connect reply
2012-08-07 11:56:48.215667 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5
ts=11917881 stream_id=0x0] len=4
2012-08-07 11:56:48.215667 [DEBUG] rtmp.c:945 Set window size: 131072 bytes

Ngrep on the traffic shows the following last packets starting with
connect: (192.168.178.103 is the PC, 192.168.178.220 is Freeswitch)
T 192.168.178.103:48179 -> 192.168.178.220:1935 [AP]
..............................................................................................K........connect.?..........app.....flashVer...LNX
11,2,202,236..swfUrl..,http://my.domain.com/flex/freeswitch.swf..tcUrl...rtmp:/./my.domain.com..fpad....capabilities. at m........audioCodecs.@.........videoCodecs. at o........videoFunction.?.........pageUrl....http://my.domain.com/flex/freeswitch.html#..objectEncoding............
##
T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
............
#
T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
....
#
T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
............
#
T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
....
#
T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
............
#
T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
.....
#
T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
............
#
T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
......
#
T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
............
##
T 192.168.178.103:48179 -> 192.168.178.220:1935 [AP]
...9............
Then traffic stops.

The packet freeswitch answeres after the connect requests is (from
wireshark):
2.109254    192.168.178.220    192.168.178.220    1935   
192.168.178.103    36804    RTMP    Unknown (0x7) | Unknown (0xf3) |
Unknown (0x21) | Unknown (0x50) | Unknown (0xff)[Packet size limited
during capture]



With kind regards
Peter


From miha at softnet.si  Tue Aug  7 14:33:31 2012
From: miha at softnet.si (Miha)
Date: Tue, 07 Aug 2012 12:33:31 +0200
Subject: [Freeswitch-users] upadte FS (make current)
In-Reply-To: <5020EC2A.2080302@softnet.si>
References: <1FFF97C269757C458224B7C895F35F1513F759@cantor.std.visionutv.se>
	<5020EC2A.2080302@softnet.si>
Message-ID: <5020EEFB.6090907@softnet.si>

@Peter,

I runed

fsctl reclaim_mem


and did not see any big difference in top (I guess I did not see any 
difference :) )

If I make this and see memory that FS is using I can not see any change:

[root at fs1 ~]# cat /proc/`ps aux | grep freeswitch | grep -v grep | awk 
'{print $2}'`/smaps | egrep "(heap)" -A 7
0ebfb000-0f96d000 rw-p 0ebfb000 00:00 0                                  
[heap]
Size:             13768 kB
Rss:              13528 kB
Shared_Clean:         0 kB
Shared_Dirty:         0 kB
Private_Clean:        0 kB
Private_Dirty:    13528 kB
Swap:                 0 kB
[root at fs1 ~]# cat /proc/`ps aux | grep freeswitch | grep -v grep | awk 
'{print $2}'`/smaps | egrep "(heap)" -A 7
0ebfb000-0f96d000 rw-p 0ebfb000 00:00 0                                  
[heap]
Size:             13768 kB
Rss:              13528 kB
Shared_Clean:         0 kB
Shared_Dirty:         0 kB
Private_Clean:        0 kB
Private_Dirty:    13528 kB
Swap:                 0 kB
So I guess is it not FS problem, I am right?

Thanks!

Regards,
Miha
On 8/7/2012 12:21 PM, Miha wrote:
> Hi Peter,
>
> fsctl reclaim_mem this will not harm any call and etc?
>
> So if freeswitch does not leak memory what should I see in top (memory will not drop?)?
>
> Thanks!
> Miha
>
>
>
> On 8/7/2012 11:48 AM, Peter Olsson wrote:
>> It's always hard to say what is a memory leak and what's not. FS uses memory pools internally, which sometime might make it look like it leaks memory, but in the end FS will reuse that memory. You can also run "fsctl reclaim_mem" in the console or fs_cli, and see if that makes the memory usage drop.
>>
>> /Peter
>>
>> -----Ursprungligt meddelande-----
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
>> Skickat: den 7 augusti 2012 11:14
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] upadte FS (make current)
>>
>> On 8/7/2012 10:54 AM, Peter Olsson wrote:
>>> Usually it's no problem. However, I always recommend to have a test
>>> server to run basic tests before going into production. There have
>>> been a couple of memory leaks fixed, but I think that was before your
>>> current version - but I don't remember all the commits from my head..
>>> :)
>>>
>>> I would also suggest to read through the git commit log, to get an idea of what has been done.
>>>
>>> /Peter
>>>
>>>
>>> -----Ursprungligt meddelande-----
>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
>>> Skickat: den 7 augusti 2012 10:37
>>> Till: FreeSWITCH Users Help
>>> ?mne: [Freeswitch-users] upadte FS (make current)
>>>
>>> Hi,
>>>
>>> I am running on production server FS (runs ok). I would like to upgrade it to latest git.
>>> Can I expect any problems? I not using any special module (just mod_rad_auth, for cdr mod_xml_cdr).
>>>
>>> Is it better to first test everything on test server and than make upgrade?
>>>
>>> My FS is now up for 13 days. I am looking at memory for a few days and noticing that free memory is shrinking (slowly). Was there any problem with memory leak? :
>>>
>>> top - 09:46:46 up 13 days, 12:32,  3 users,  load average: 0.00, 0.04, 0.05
>>> Tasks: 159 total,   1 running, 158 sleeping,   0 stopped,   0 zombie
>>> Cpu(s):  0.6%us,  0.4%sy,  0.0%ni, 99.0%id,  0.0%wa,  0.0%hi, 0.0%si, 0.0%st
>>> Mem:   4046636k total,  3356340k used,   690296k free,   357296k buffers
>>> Swap:  6094840k total,        0k used,  6094840k free,  2629280k cached
>>>
>>> My version: FreeSWITCH Version
>>> 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5
>>> (1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul 2012 15:18:27 Z)
>>>
>>> Thanks!
>>>
>>> Miha
>>>
>>> ______________________________________________________________________
>>> ___ Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>> rs
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> ______________________________________________________________________
>>> ___ Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>> rs
>>> http://www.freeswitch.org
>>>
>> @Peter thanks for your quick answer:)
>>
>> I readed jira postes and noticed that this was befor my git version (as you said).
>> If there was leak problem (I do not know if it is, just noticing that free memory is droping), would FS be stable for 13 days or would this bug be present sooner (memory leak)?
>>
>> regards,
>> Miha
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>  
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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:5020daa032762709953773!
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From hkalyoncu at gmail.com  Tue Aug  7 14:40:59 2012
From: hkalyoncu at gmail.com (huseyin kalyoncu)
Date: Tue, 7 Aug 2012 13:40:59 +0300
Subject: [Freeswitch-users] freeswitch_licence_server problem
In-Reply-To: <CAPf+86UZZT2h87frz=gC0hfJbBq7Qo7Rf_AjbMP_5q79NXq9XA@mail.gmail.com>
References: <CAB+ia=2DkqDXvPZEvRTWqVYF=1i3S4ViryEbwiJM9FxrtM9=Wg@mail.gmail.com>
	<CAPf+86UZZT2h87frz=gC0hfJbBq7Qo7Rf_AjbMP_5q79NXq9XA@mail.gmail.com>
Message-ID: <CAPf+86UwNhbMv9O-Vxp1AaGOKZ+vYq1mmHJj_=HiRsmj4OCgEg@mail.gmail.com>

hi,
problem solved! it was easy after all.

if your network interface name is non-standard (not ethX etc.) rename it to
standard ethX name.
after reboot you can install g729 license.

regards
huseyin

On Fri, Aug 3, 2012 at 9:41 AM, huseyin kalyoncu <hkalyoncu at gmail.com>wrote:

> i have the same problem. but i cant get any response from
> consulting at freeswitch.org
>
> anybody solved this kind of problem before?
>
> thanks
> huseyin
>
>
> On Wed, Aug 1, 2012 at 2:40 PM, Ale <ale975 at gmail.com> wrote:
>
>> Hello,
>>
>> short: after installation fs show "Can't contact licence server." and
>> freeswitch_licence_server in console show "Unrecognised resource
>> G.729A/0".
>>
>> I've purchased a license, downloaded 194 installer, mod_com_g729.so is
>> in lib64/freeswitch/mod dir, and license file without licenze.zip is
>> in /etc/freeswitch.
>> Freeswitch run as user freeswitch user on a centos.
>> Where i unload mod_g729 and load the mod_com, server correctly starts.
>> I also try to kill the server,
>> and manually start it a root user, but nothing change.
>>
>> could someone give me any hints?
>> Thanks alessandro
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From alex at thewinelake.com  Tue Aug  7 15:25:05 2012
From: alex at thewinelake.com (Alex)
Date: Tue, 07 Aug 2012 12:25:05 +0100
Subject: [Freeswitch-users] One Way Audio
In-Reply-To: <501FD0F5.2000600@thewinelake.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<50194496.10804@bestnet.kharkov.ua>
	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>
	<501A5AF4.9020008@bestnet.kharkov.ua>
	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>
	<501A6085.2020308@bestnet.kharkov.ua>
	<CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>
	<501BAE64.3020304@bestnet.kharkov.ua>
	<CAJUJwtgHFyrk7ybxAmH2xmaz=d=OgCy5uHOmGhnyOK3QpdRjRw@mail.gmail.com>
	<CAPBKugskuRA2FgE6H+fUsHrU4+N-E+2erantqShSKn7H6Qw99g@mail.gmail.com>
	<CAJUJwti68vmSDFpAkwk08otnJyWHkOWJ6qRJ=AcStmtW=RT04Q@mail.gmail.com>
	<501BD835.3060308@bestnet.kharkov.ua>
	<501FC533.60708@thewinelake.com>
	<CAF+3j-v7LudQh6T7s7WyCOuMjnBkPHR3uD9PdP_LNc=VV7k00g@mail.gmail.com>
	<501FC7E8.8030609@thewinelake.com>
	<501FD0F5.2000600@thewinelake.com>
Message-ID: <5020FB11.3090408@thewinelake.com>

Looks like this is specific to LinkSys SPA942. Other handsets (eg. 
Grandstream) are fine.
It's intermittent. Wondering if it's negotiating some "unlucky" port 
addresses.
> Getting an occasional one-way audio problem.
>
> One thing I was wondering about is this:
>
> switch_rtp.c:3240 Auto Changing port from 86.174.119.10:16432 to 86.174.119.10:61384
>
> Can one somehow disable this feature as part of diagnostics?
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2196 / Virus Database: 2437/5181 - Release Date: 08/06/12
>
>



From ben at langfeld.co.uk  Tue Aug  7 15:56:30 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Tue, 7 Aug 2012 12:56:30 +0100
Subject: [Freeswitch-users] Make outbound call
In-Reply-To: <CANJXrA9nxtDTLUuOZRi270Dmy=-s1FowXBXfAtjgw73t7eQ62Q@mail.gmail.com>
References: <CAO-yT6hrFqT_hKaqO41GWppsVqHbcSBwkNbMUg1R1Ue03JLAZw@mail.gmail.com>
	<CANJXrA9nxtDTLUuOZRi270Dmy=-s1FowXBXfAtjgw73t7eQ62Q@mail.gmail.com>
Message-ID: <CAAyX+KHGARkzV9t58zjL_xGnRA_5CBq6M-9eF+a2WNwB_fk-Pw@mail.gmail.com>

You have to use inbound event socket.

Inbound: client connects to freeswitch
Outbound: freeswitch connects to client

In outbound mode, you only get a connection to the client when you
have an active call. In order to originate arbitrary calls, you need
to make a connection from your client in to FreeSWITCH.

Regards,
Ben Langfeld


On 7 August 2012 09:16, Prevail Magid <m.prevail at gmail.com> wrote:
> You can do it by several way.
>
> For example:
>
> sendAsyncCommand( channel, "bgapi originate user/1001 777");
> http://wiki.freeswitch.org/wiki/Mod_commands#originate
>
>
> 2012/8/7 Khue Nguyen Minh <khuenm at vega.com.vn>
>>
>> Hi all,
>>
>> I am using event socket of FreeSWITCH in "outbound" mode. From my
>> application, I want request FreeSWITCH make outbound call to a SIP client.
>> How I can do it, please guide me?
>>
>> Best regards,
>> Khue Nguyen.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From trob at freemail.hu  Tue Aug  7 17:01:33 2012
From: trob at freemail.hu (=?ISO-8859-2?Q?T=F3th_R=F3bert?=)
Date: Tue, 07 Aug 2012 15:01:33 +0200
Subject: [Freeswitch-users] Run on Windows without install
Message-ID: <502111AD.4060909@freemail.hu>

Hello

I would like to use Freeswitch on Windows without installing by the instaler.
I build FS from git, but it won't run without a previous install.

So i would like to know what does the installer do.
Or is there an installer what does the necessary settings, but not copy the files?

thx
Robert
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From peter.olsson at visionutveckling.se  Tue Aug  7 17:19:56 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 7 Aug 2012 13:19:56 +0000
Subject: [Freeswitch-users] Run on Windows without install
Message-ID: <1FFF97C269757C458224B7C895F35F1513FA9A@cantor.std.visionutv.se>

It doesn't do anything special at all as far as I know.

It's enough to copy the files and just start freeswitch.exe (or FreeSwitchConsole.exe if you're in VS2010).

If you want to register as a service you just use "-install" to install the service. And also, if you just copy the file to another computer, you need to make sure that vcredist for your compiler is installed.

If this isn't working for you, please get back with info about what kind of problems that occur.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r T?th R?bert
Skickat: den 7 augusti 2012 15:02
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Run on Windows without install

Hello

I would like to use Freeswitch on Windows without installing by the instaler.
I build FS from git, but it won't run without a previous install.

So i would like to know what does the installer do.
Or is there an installer what does the necessary settings, but not copy the files?

thx
Robert
!DSPAM:5021101932769323245192!
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From trob at freemail.hu  Tue Aug  7 17:37:36 2012
From: trob at freemail.hu (=?ISO-8859-1?Q?T=F3th_R=F3bert?=)
Date: Tue, 07 Aug 2012 15:37:36 +0200
Subject: [Freeswitch-users] Run on Windows without install
In-Reply-To: <502119A4.6010001@freemail.hu>
References: <502119A4.6010001@freemail.hu>
Message-ID: <50211A20.8040407@freemail.hu>



Thanks, the solution was installing the redistributable...


On 8/7/2012 11:48 AM, Peter Olsson wrote:

> It doesn't do anything special at all as far as I know.
>
> It's enough to copy the files and just start freeswitch.exe (or FreeSwitchConsole.exe if you're in VS2010).
>
> If you want to register as a service you just use "-install" to install the service. And also, if you just copy the file to another computer, you need to make sure that vcredist for your compiler is installed.
>
> If this isn't working for you, please get back with info about what kind of problems that occur.
>
> /Peter
>
>
> Fr?n:freeswitch-users-bounces at lists.freeswitch.org  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>  [mailto:freeswitch-users-bounces at lists.freeswitch.org  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>] F?r T?th R?bert
> Skickat: den 7 augusti 2012 15:02
> Till:freeswitch-users at lists.freeswitch.org  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> ?mne: [Freeswitch-users] Run on Windows without install
>
> Hello
>
> I would like to use Freeswitch on Windows without installing by the instaler.
> I build FS from git, but it won't run without a previous install.
>
> So i would like to know what does the installer do.
> Or is there an installer what does the necessary settings, but not copy the files?
>
> thx
> Robert



From mgg at giagnocavo.net  Tue Aug  7 17:54:56 2012
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 7 Aug 2012 13:54:56 +0000
Subject: [Freeswitch-users] One Way Audio
In-Reply-To: <5020FB11.3090408@thewinelake.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<50194496.10804@bestnet.kharkov.ua>
	<CAJUJwthmd=q7VJhHnL=01p=GxDftdij9R_k646Jc5fz+RJd8cA@mail.gmail.com>
	<501A5AF4.9020008@bestnet.kharkov.ua>
	<CAJUJwtg+d_A16mtyVCwqabSb2FjifAbAj06hLAyrB0QSOyOTyg@mail.gmail.com>
	<501A6085.2020308@bestnet.kharkov.ua>
	<CAJUJwthqUVA4L02uEpGJRUfBOY2D3ixYJWLJNm-GCvU55UpuXg@mail.gmail.com>
	<501BAE64.3020304@bestnet.kharkov.ua>
	<CAJUJwtgHFyrk7ybxAmH2xmaz=d=OgCy5uHOmGhnyOK3QpdRjRw@mail.gmail.com>
	<CAPBKugskuRA2FgE6H+fUsHrU4+N-E+2erantqShSKn7H6Qw99g@mail.gmail.com>
	<CAJUJwti68vmSDFpAkwk08otnJyWHkOWJ6qRJ=AcStmtW=RT04Q@mail.gmail.com>
	<501BD835.3060308@bestnet.kharkov.ua>	<501FC533.60708@thewinelake.com>
	<CAF+3j-v7LudQh6T7s7WyCOuMjnBkPHR3uD9PdP_LNc=VV7k00g@mail.gmail.com>
	<501FC7E8.8030609@thewinelake.com>	<501FD0F5.2000600@thewinelake.com>
	<5020FB11.3090408@thewinelake.com>
Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B612F913B2@BLUPRD0711MB413.namprd07.prod.outlook.com>

Try getting a full SIP + RTP capture. That message should be for detecting the proper port, but sometimes I think FS goes symmetric when it should not, which causes one-way audio.

-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Alex
Sent: Tuesday, August 07, 2012 5:25 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] One Way Audio

Looks like this is specific to LinkSys SPA942. Other handsets (eg. 
Grandstream) are fine.
It's intermittent. Wondering if it's negotiating some "unlucky" port addresses.
> Getting an occasional one-way audio problem.
>
> One thing I was wondering about is this:
>
> switch_rtp.c:3240 Auto Changing port from 86.174.119.10:16432 to 
> 86.174.119.10:61384
>
> Can one somehow disable this feature as part of diagnostics?
>
>
>
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2196 / Virus Database: 2437/5181 - Release Date: 
> 08/06/12
>
>


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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




From vaad.fabi at gmail.com  Tue Aug  7 18:53:52 2012
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Tue, 07 Aug 2012 17:53:52 +0300
Subject: [Freeswitch-users] RTMP subclass rtmp::unregister event variable
	missing
In-Reply-To: <34260552.post@talk.nabble.com>
References: <34260552.post@talk.nabble.com>
Message-ID: <50212C00.7070806@gmail.com>

  Hi all,

I am try to realize simple lua event socket script to check rtmp mod 
events. But found some issue according to rtmp events -  for subclass  
rtmp::register we can get 'USER' variable, but for  subclass 
rtmp::unregister  there is no variable USER and some others.
Please explain how to get user name on rtmp::unregister event.  Dumps 
for all subclases are below.



Event-Name: CUSTOM
Core-UUID: 211bfd74-e09c-11e1-b583-492137a6e73f
FreeSWITCH-Hostname: fs-flashphone.XXX.net
FreeSWITCH-Switchname: fs-flashphone.XXX.net
FreeSWITCH-IPv4: 193.158.000.104
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2012-08-07%2018%3A40%3A56
Event-Date-GMT: Tue,%2007%20Aug%202012%2014%3A40%3A56%20GMT
Event-Date-Timestamp: 1344350456515761
Event-Calling-File: mod_rtmp.c
Event-Calling-Function: rtmp_add_registration
Event-Calling-Line-Number: 1045
RTMP-Session-ID: 235bae22-e09c-11e1-b587-492137a6e73f
RTMP-Flash-Version: WIN%2011,3,300,268
RTMP-SWF-URL: http%3A//193.158.000.104/flash/freeswitch.swf
RTMP-TC-URL: rtmp%3A//193.158.000.104
RTMP-Page-URL: http%3A//193.158.000.104/flash/freeswitch.html
RTMP-Profile: default
Network-Port: 3896
Network-IP: 92.116.000.140
RTMP-Profile: default
User: test04
Domain: 193.158.000.104




Event-Name: CUSTOM
Core-UUID: 211bfd74-e09c-11e1-b583-492137a6e73f
FreeSWITCH-Hostname: fs-flashphone.XXX.net
FreeSWITCH-Switchname: fs-flashphone.XXX.net
FreeSWITCH-IPv4: 193.158.000.104
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2012-08-07%2018%3A41%3A24
Event-Date-GMT: Tue,%2007%20Aug%202012%2014%3A41%3A24%20GMT
Event-Date-Timestamp: 1344350484383919
Event-Calling-File: mod_rtmp.c
Event-Calling-Function: rtmp_clear_reg_auth
Event-Calling-Line-Number: 1076
RTMP-Session-ID: 235bae22-e09c-11e1-b587-492137a6e73f
RTMP-Flash-Version: WIN%2011,3,300,268
RTMP-SWF-URL: http%3A//193.158.000.104/flash/freeswitch.swf
RTMP-TC-URL: rtmp%3A//193.158.000.104
RTMP-Page-URL: http%3A//193.158.000.104/flash/freeswitch.html
RTMP-Profile: default
Network-Port: 3896
Network-IP: 92.116.000.140
RTMP-Profile: default

-- 
Best Regards,
Vadim F.



From jack at livecall.com  Tue Aug  7 19:51:57 2012
From: jack at livecall.com (Jack)
Date: Tue, 07 Aug 2012 08:51:57 -0700
Subject: [Freeswitch-users] Problem with flex client connecting to
	mod_rtmp
In-Reply-To: <5020CDB9.3060102@telefaks.de>
References: <502064F5.2040702@telefaks.de> <50209901.10107@livecall.com>
	<5020CDB9.3060102@telefaks.de>
Message-ID: <5021399D.2030501@livecall.com>

Peter,
   Here is my rtmp.conf.xml:

<configuration name="rtmp.conf" description="RTMP Endpoint">
   <profiles>
     <profile name="default">
         <settings>
             <param name="bind-address" value="0.0.0.0:1935" />
   <param name="context" value="public" />
             <param name="dialplan" value="XML" />
             <!-- If this is set to true, no unauthenticated inbound 
calls will be allowed -->
             <param name="auth-calls" value="true" />
             <!-- How much time should the clients buffer the media 
stream (miliseconds) -->
             <param name="buffer-len" value="50" />
             <!-- Sets the maximum size of outbound RTMP chunks -->
             <param name="chunksize" value="512" />
         </settings>
     </profile>
   </profiles>
</configuration>

It is important to specify the context and then make sure you have a 
condition in your corresponding dial plan that will catch your call.

Jack



On 8/7/2012 1:11 AM, Peter Steinbach wrote:
> Hello Jack,
>
> here are some answers to your your questions:
>
>> Make sure you use a fully qualified username not just the extension
>> 1001 at xxx.xxx.xxx.xxx >the XXX would be the IP of your FreeSwitch Server.
> I am still in the "connect" state, I am not logging in yet. The webpage
> tells me "Connecting... " and waits forever
>
>> did you configure your rtmp.conf.xml and have a matching context in
> your dial plan?
> I am still in the "connect" state
>
> In the web page make sure these two vars are set to YOUR Freeswitch
> server ip: var rtmpIPURL =
>> "rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx";
> I did not have a rtmpIP var, so I added it to the code, but no change.
>
> In fact as I can see while grepping the network traffic, the Flex app
> and Freeswicth are negociating their protocol and freeswitch logs a
> "Sent connect reply", but this is very diferent from what is sent from
> the conference.freeswitch.org server.
>
> Best regards
> Peter
>
>
> On 08/07/12 06:26, Jack wrote:
>> Hi Peter,
>> Make sure you use a fully qualified username not just the extension
>> 1001 at xxx.xxx.xxx.xxx
>> the XXX would be the IP of your FreeSwitch Server.
>>
>> did you configure your rtmp.conf.xml and have a matching context in your
>> dial plan?
>>
>> In the web page make sure these two vars are set to YOUR Freeswitch
>> server ip:
>> var rtmpIPURL = "rtmp://xxx.xxx.xxx.xxx/phone";
>>    var rtmpIP="xxx.xxx.xxx.xxx";
>>
>> jack
>>
>> On 8/6/2012 5:44 PM, Peter Steinbach wrote:
>>> Hello,
>>>
>>> today I tried the flex client with the mod_rtmp implementation.
>>> On a brandnew freeswitch I installed mod_rtmp and copied the flex
>>> direrectory to a web server and loaded the web page.
>>>
>>> However the flex client does not connect:
>>> Here's my freeswitch log
>>> 2012-08-07 02:32:03.585671 [NOTICE] mod_rtmp.c:744 New RTMP session
>>> [f35a192c-0d02-4320-89e9-775a3573ee25]
>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:702 Sent handshake response
>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:727 Done with handshake
>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14
>>> ts=0 stream_id=0x0] len=342
>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE
>>> for connect
>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1
>>> stream_id=0x0] len=4
>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5
>>> stream_id=0x0] len=4
>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6
>>> stream_id=0x0] len=5
>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4
>>> stream_id=0x0] len=6
>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
>>> stream_id=0x0] len=201
>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
>>> stream_id=0x0] len=61
>>> 2012-08-07 02:32:03.685670 [NOTICE] rtmp_sig.c:121 Sent connect reply
>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5
>>> ts=12247364 stream_id=0x0] len=4
>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
>>>
>>> If I change the rtmp_url to 'rtmp://conference.freeswitch.org/phone',
>>> flex does connect.
>>>
>>> But in my freeswitch this fails. I have traced the network traffic and I
>>> can see that there is information flow between freeswitch and the client
>>> on connect request.
>>>
>>> Somebody has an idea where to look further?
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

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From krice at freeswitch.org  Tue Aug  7 20:03:34 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 7 Aug 2012 11:03:34 -0500
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
Message-ID: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>

The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!

Get your copy today at http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2!

Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
continue to fix bugs and security issues. giving you a stable platform for
at least one year.

Grab it today!

The FreeSWITCH Team
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From mitch.capper at gmail.com  Tue Aug  7 20:06:02 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 7 Aug 2012 09:06:02 -0700
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
References: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
Message-ID: <CALn233qBPYhVN2byHUntarpuSLm+Rp9j8adMaJ4R7nd1=mDS_A@mail.gmail.com>

Awesome!

~Mitch

On Tue, Aug 7, 2012 at 9:03 AM, Ken Rice <krice at freeswitch.org> wrote:
> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!
>
> Get your copy today at http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2
> !
>
> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
> continue to fix bugs and security issues. giving you a stable platform for
> at least one year.
>
> Grab it today!
>
> The FreeSWITCH Team
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From sdevoy at bizfocused.com  Tue Aug  7 20:16:03 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 7 Aug 2012 12:16:03 -0400
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
References: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
Message-ID: <242201cd74b7$f17e0ee0$d47a2ca0$@bizfocused.com>

I hate to show my Unix illiteracy, but is that the same thing I get with
make current?

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Tuesday, August 07, 2012 12:04 PM
To: FreeSWITCH Users Help; freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2

 

The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!

 

Get your copy today at http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2
! 

 

Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
continue to fix bugs and security issues. giving you a stable platform for
at least one year.

 

Grab it today!

 

The FreeSWITCH Team 

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From krice at freeswitch.org  Tue Aug  7 20:22:28 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 7 Aug 2012 11:22:28 -0500
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <242201cd74b7$f17e0ee0$d47a2ca0$@bizfocused.com>
References: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
	<242201cd74b7$f17e0ee0$d47a2ca0$@bizfocused.com>
Message-ID: <CAPcNkdfEux6a_NrpWtDrFYjTsZdvrfipHekyaWjP4FpHix32Bg@mail.gmail.com>

Right now if you make current you will basically get the release version
however keep in mind that you will stay on the development branch... stay
tuned for more info on this coming soon for those that want to track stable
from git!

K

On Tue, Aug 7, 2012 at 11:16 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> I hate to show my Unix illiteracy, but is that the same thing I get with
> make current?****
>
> ** **
>
> Sean****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice
> *Sent:* Tuesday, August 07, 2012 12:04 PM
> *To:* FreeSWITCH Users Help; freeswitch-dev at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Announcing FreeSWITCH 1.2****
>
> ** **
>
> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!****
>
> ** **
>
> Get your copy today at
> http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 ! ****
>
> ** **
>
> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
> continue to fix bugs and security issues. giving you a stable platform for
> at least one year.****
>
> ** **
>
> Grab it today!****
>
> ** **
>
> The FreeSWITCH Team ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gabe at gundy.org  Tue Aug  7 20:25:23 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 7 Aug 2012 11:25:23 -0500
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
References: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
Message-ID: <CAMwZ-tL1D2+brGEZKngQgZGeTVop96bzBrj0pLneQnCEgMC6kA@mail.gmail.com>

On Tue, Aug 7, 2012 at 11:03 AM, Ken Rice <krice at freeswitch.org> wrote:
> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!

Strong work everyone! Huge milestone!


Gabe


From gabe at gundy.org  Tue Aug  7 20:32:50 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 7 Aug 2012 11:32:50 -0500
Subject: [Freeswitch-users] Calling all Python users @ ClueCon
Message-ID: <CAMwZ-tKg64MBE8YfWNhyHpoXFnh1pcMF07M0qY3CvFkETYaqYw@mail.gmail.com>

Esteemed Pythonistas at ClueCon,

Any Python users want to find an evening to talk shop and hang out at
ClueCon this year? We planned on this last year, but we dropped the
ball :) Anyway, let me know!

Best,
Gabe


From anatoliy at kounitskiy.com  Tue Aug  7 20:48:08 2012
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Tue, 7 Aug 2012 19:48:08 +0300
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
References: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
Message-ID: <97D36ACD-601D-4D15-8BA0-E44D11BBC9A3@kounitskiy.com>

Time to start testing & planning migrations :D

Thanks!

On Aug 7, 2012, at 7:03 PM, Ken Rice <krice at freeswitch.org> wrote:

> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!
> 
> Get your copy today at http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 ! 
> 
> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will continue to fix bugs and security issues. giving you a stable platform for at least one year.
> 
> Grab it today!
> 
> The FreeSWITCH Team 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jason at jg-solutions.com  Tue Aug  7 20:47:44 2012
From: jason at jg-solutions.com (Jason A. Gould)
Date: Tue, 7 Aug 2012 16:47:44 +0000
Subject: [Freeswitch-users] One way audio only outbound to only AT&T mobile
	numbers
Message-ID: <1FFCB7133597F34F9F61042A443ACECF8195AD44@TGH-MX2.thegouldhouse.net>

FreeSwitch noob here.

 

I currently have a mostly working setup here but I have one rather major
problem. I can send and receive calls from most numbers perfectly, the only
numbers that seem to be giving me problems are AT&T mobile numbers. Inbound
AT&T calls work fine, but outbound AT&T calls only have inbound audio, audio
from my switch to AT&T is lost. Naturally most of the mobile numbers I need
to call are AT&T.

 

According to my VoIP provider the problem is I'm sending out asynchronous
RTP (my send and receive ports are different) and that the media gateway on
the other end only allows synchronous RTP so my send and receive ports need
to be the same. Unfortunately I can't seem to find any setting in FS to
enable sync RTP. I found this under RTP issues:

 

<!-- This will use the async timer (this one you can also set to none if you
don't want asynchronous RTP traffic) --> 

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

 

But setting rtp-timer-name to none still uses different RTP ports for the
outbound traffic.

 

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From lists.jj at googlemail.com  Tue Aug  7 20:03:54 2012
From: lists.jj at googlemail.com (Johannes Jakob)
Date: Tue, 7 Aug 2012 18:03:54 +0200
Subject: [Freeswitch-users] Call Diversion/CF (302 Moved Temporarily) not
	working
Message-ID: <CAO3-fvaKVtN6HewXzSGRFK-Ot9uCcRm_z1Dsf=yD7ETT6oQE3w@mail.gmail.com>

Hi,

My setup looks like this:

User A (1001), User B (1002) and User C (1003) are registered to
13.13.13.66 via OpenSIPs Proxy 222.222.222.222.
User B activated unconditional call forwarding on his phone to User C.
Now User A (or any external caller) calls User B.

Until last week User A would then be able to talk to User C - the
expected behaviour.
On Saturday, I updated the freeswitch from version 9fe08675a1 to
d1c3f910a6 to get and test some of Steve's fax changes.
Since this upgrade, Call-Forwarding / Call Diversion is broken on a
level, that I don't understand.


User A: INVITE 1002 to B's phone
=> User B: 302 Moved Temporarily (Contact: <sip:1003 at sip.isp.net;user=phone>)
=> FS: ACK
=> FS: INVITE 1003 to *B*'s phone!
=> User B: 404 Not Found
=> FS: ACK
=> User A gets signalled the 404 Not Found.

packet trace below.




I'm aware of
http://jira.freeswitch.org/browse/FS-724
http://jira.freeswitch.org/browse/FS-821

but those didn't help me solve my issue:

internal.xml:		<param name="apply-nat-acl" value="nat.auto"/>
internal.xml:		<param name="aggressive-nat-detection" value="false"/>

I added aggressive-nat-detection=false yesterday, before it wasn't in
the config at all.
I also tried removing the nat.auto acl temporarily, but that didn't help either.


Yesterday afternoon I updated to git c3de9637af, did several
bootstrappings, cleans and makes... no change.

So today I tried to downgrade to 9fe08675a1, the version I had running
until Friday and that was working fine... well, it doesn't now. So I
can't even _prove_ it was working before :(



Can somebody please point me in the right direction?




Any hint in the right direction is much appreciated!



Thanks!





======================================================================================

SIP Trace (tcpdump):


16:38:58.950321 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472
E....[ .>..;^...
	.......D.fINVITE sip:1002 at 137.137.137.245:7390;line=ydwzxzpg SIP/2.0
Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285>
Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0
Via: SIP/2.0/UDP
13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB
Max-Forwards: 28
From: "User A" <sip:1001 at sip.isp.net>;tag=3r8K7r5Hppe6N
To: <sip:1002 at 137.137.137.245:7390;line=ydwzxzpg>
Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd
CSeq: 31839105 INVITE
Contact: <sip:mod_sofia at 13.13.13.66:5060>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 442
P-Key-Flags: keys="3"
X-AUTH-IP: 137.137.137.245
X-FS-Support: update_display,send_info
Remote-Party-ID: "User A"
<sip:1001 at sip.isp.net>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1344322582 1344322583 IN IP4 13.13.13.66
s=FreeSWITCH
c=IN IP4 13.13.13.66
t=0 0
m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101 13
a=rtpmap:98 G7221/32000
a=fmtp:98 bitrate=48000
a=rtpmap:99 G7221/16000
a=fmtp:99 bitrate=32000
a=rtpmap:100 iLBC/8000
a=fmtp:100 mode=30
a=rtpmap:101 teleph
16:38:58.950328 IP 222.222.222.222 > 172.16.12.210: udp
E....[..>.4.^...
	..one-event/8000
a=fmtp:101 0-16
a=ptime:20
m=video 27936 RTP/AVP 34 102
a=rtpmap:34 H263/90000
a=rtpmap:102 H264/90000

16:38:58.979148 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 623
E..... at .@.
.
	..^........w8$SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0
Via: SIP/2.0/UDP
13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB
Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285>
From: "User A" <sip:1001 at sip.isp.net>;tag=3r8K7r5Hppe6N
To: <sip:1002 at 137.137.137.245:7390;line=ydwzxzpg>;tag=mzey6xfxhz
Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd
CSeq: 31839105 INVITE
Contact: <sip:1003 at sip.isp.net;user=phone>
Diversion: <sip:1002 at 137.137.137.245:7390;line=ydwzxzpg>;reason="unconditional"
Content-Length: 0


16:38:58.979954 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 358
.^.... at .>.
	.......n.fACK sip:1002 at 137.137.137.245:7390;line=ydwzxzpg SIP/2.0
Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0
From: "User A" <sip:1001 at sip.isp.net>;tag=3r8K7r5Hppe6N
Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd
To: <sip:1002 at 137.137.137.245:7390;line=ydwzxzpg>;tag=mzey6xfxhz
CSeq: 31839105 ACK
Max-Forwards: 70
Content-Length: 0


16:38:58.985405 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472
E....\ .>..:^...
	.......ZgIINVITE sip:1003 at 137.137.137.245:7390;user=phone SIP/2.0
Record-Route: <sip:222.222.222.222;lr;ftag=411c9KpNKZ4rH;did=e81.20343ac4>
Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0
Via: SIP/2.0/UDP
13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ
Max-Forwards: 28
From: "User A" <sip:1001 at sip.isp.net>;tag=411c9KpNKZ4rH
To: <sip:1002 at 137.137.137.245:7390;line=ydwzxzpg>
Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd
CSeq: 31839105 INVITE
Contact: <sip:mod_sofia at 13.13.13.66:5060>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 466
P-Key-Flags: keys="3"
X-AUTH-IP: 137.137.137.245
X-FS-Support: update_display,send_info
Remote-Party-ID: "User A"
<sip:1001 at sip.isp.net>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1344322582 1344322584 IN IP4 13.13.13.66
s=FreeSWITCH
c=IN IP4 13.13.13.66
t=0 0
m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101
a=rtpmap:98 G7221/32000
a=fmtp:98 bitrate=48000
a=rtpmap:99 G7221/16000
a=fmtp:99 bitrate=32000
a=rtpmap:100 iLBC/8000
a=fmtp:100 mode=30
a=rtpmap:101 telephone-e
16:38:58.985409 IP 222.222.222.222 > 172.16.12.210: udp
E....\..>.4.^...
	..vent/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
m=video 27936 RTP/AVP 34 102
a=rtpmap:34 H263/90000
a=rtpmap:102 H264/90000

16:38:59.031218 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 551
E..C.. at .@..7
	..^......../..SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0
Via: SIP/2.0/UDP
13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ
From: "User A" <sip:1001 at sip.isp.net>;tag=411c9KpNKZ4rH
To: <sip:1002 at 137.137.137.245:7390;line=ydwzxzpg>
Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd
CSeq: 31839105 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Length: 0


16:38:59.031922 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 341
E..q.. at .>...^...
	.......].mACK sip:1003 at 137.137.137.245:7390;user=phone SIP/2.0
Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0
From: "User A" <sip:1001 at sip.isp.net>;tag=411c9KpNKZ4rH
Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd
To: <sip:1002 at 137.137.137.245:7390;line=ydwzxzpg>
CSeq: 31839105 ACK
Max-Forwards: 70
Content-Length: 0


======================================================================================


From basit.engg at gmail.com  Tue Aug  7 20:54:30 2012
From: basit.engg at gmail.com (Abdul Basit)
Date: Tue, 7 Aug 2012 21:54:30 +0500
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <97D36ACD-601D-4D15-8BA0-E44D11BBC9A3@kounitskiy.com>
References: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
	<97D36ACD-601D-4D15-8BA0-E44D11BBC9A3@kounitskiy.com>
Message-ID: <CAGQG5MaG7MmnV7yR-bEXyDP3PcaW5mGOR_htBYVXsJm+oZO_9w@mail.gmail.com>

good mile stone. Good work team.


-- 
Regards,

Abdul Basit


On Tue, Aug 7, 2012 at 9:48 PM, Anatoliy Kounitskiy <anatoliy at kounitskiy.com
> wrote:

> Time to start testing & planning migrations :D
>
> Thanks!
>
> On Aug 7, 2012, at 7:03 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!
>
> Get your copy today at
> http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !
>
> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
> continue to fix bugs and security issues. giving you a stable platform for
> at least one year.
>
> Grab it today!
>
> The FreeSWITCH Team
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From red.rain.seven at gmail.com  Tue Aug  7 20:56:46 2012
From: red.rain.seven at gmail.com (Henry Huang)
Date: Tue, 7 Aug 2012 09:56:46 -0700
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAPcNkdfEux6a_NrpWtDrFYjTsZdvrfipHekyaWjP4FpHix32Bg@mail.gmail.com>
References: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
	<242201cd74b7$f17e0ee0$d47a2ca0$@bizfocused.com>
	<CAPcNkdfEux6a_NrpWtDrFYjTsZdvrfipHekyaWjP4FpHix32Bg@mail.gmail.com>
Message-ID: <CAP_G395toBY=TeuA_4JBg2zmVkr6pV6Z8xNn4WtR0NVf6JynHg@mail.gmail.com>

Is there a What's new page some where?

Henry

On Tue, Aug 7, 2012 at 9:22 AM, Ken Rice <krice at freeswitch.org> wrote:

> Right now if you make current you will basically get the release version
> however keep in mind that you will stay on the development branch... stay
> tuned for more info on this coming soon for those that want to track stable
> from git!
>
> K
>
> On Tue, Aug 7, 2012 at 11:16 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>
>> I hate to show my Unix illiteracy, but is that the same thing I get with
>> make current?****
>>
>> ** **
>>
>> Sean****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice
>> *Sent:* Tuesday, August 07, 2012 12:04 PM
>> *To:* FreeSWITCH Users Help; freeswitch-dev at lists.freeswitch.org
>> *Subject:* [Freeswitch-users] Announcing FreeSWITCH 1.2****
>>
>> ** **
>>
>> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!****
>>
>> ** **
>>
>> Get your copy today at
>> http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 ! ****
>>
>> ** **
>>
>> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
>> continue to fix bugs and security issues. giving you a stable platform for
>> at least one year.****
>>
>> ** **
>>
>> Grab it today!****
>>
>> ** **
>>
>> The FreeSWITCH Team ****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From monemran at gmail.com  Tue Aug  7 20:52:46 2012
From: monemran at gmail.com (Mohammad Emran)
Date: Tue, 7 Aug 2012 22:52:46 +0600
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAMwZ-tL1D2+brGEZKngQgZGeTVop96bzBrj0pLneQnCEgMC6kA@mail.gmail.com>
References: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
	<CAMwZ-tL1D2+brGEZKngQgZGeTVop96bzBrj0pLneQnCEgMC6kA@mail.gmail.com>
Message-ID: <447EBF98-8B6F-4B67-A3AB-989E0C1B48D3@gmail.com>

Love u guys!!!!!



Sent from my iPhone

On Aug 7, 2012, at 10:25 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Tue, Aug 7, 2012 at 11:03 AM, Ken Rice <krice at freeswitch.org> wrote:
>> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!
> 
> Strong work everyone! Huge milestone!
> 
> 
> Gabe
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From jack at livecall.com  Tue Aug  7 22:11:13 2012
From: jack at livecall.com (Jack)
Date: Tue, 07 Aug 2012 11:11:13 -0700
Subject: [Freeswitch-users] webrtc connecting to FreeSwitch
Message-ID: <50215A41.6020702@livecall.com>

Has anyone been successful calling into freeswitch with webrtc through a 
webpage?
If so , could you share your html?
Thanks,
  Jack


From drk at drkngs.net  Tue Aug  7 22:12:33 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Tue, 07 Aug 2012 11:12:33 -0700
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
Message-ID: <20120807181233.ee23e2d0@mail.tritonwest.net>

Where is the branch so I can checkout local?  
   
D:\fsnew>git branch -a
* master
  remotes/origin/FS-3432
  remotes/origin/FS-4062
  remotes/origin/HEAD -> origin/master
  remotes/origin/dingaling_video
  remotes/origin/master
  remotes/origin/smgfs
  remotes/origin/stable-test/freeswitch-1.2
  remotes/origin/swk/fs_test
  remotes/test/master
  
--Dave
      _____  

  From: Ken Rice [mailto:krice at freeswitch.org]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org], freeswitch-dev at lists.freeswitch.org
Sent: Tue, 07 Aug 2012 09:03:34 -0700
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2

The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!  

  
Get your copy today at http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !   

  
Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will continue to fix bugs and security issues. giving you a stable platform for at least one year.  

  
Grab it today!  

  
The FreeSWITCH Team       
   
 
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From krice at freeswitch.org  Tue Aug  7 22:15:50 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 7 Aug 2012 13:15:50 -0500
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <20120807181233.ee23e2d0@mail.tritonwest.net>
References: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
	<20120807181233.ee23e2d0@mail.tritonwest.net>
Message-ID: <CAPcNkdeC5Yp3ht_4KqYzrHEpVUAYuz_R9gjYi2EDmEepVqLFBQ@mail.gmail.com>

its Tagged, right now, we will be branching it shortly for tracking
patches. Please note we had a couple of good users point out some smallish
bugs so we're int he process of fixing those! so guys if you find anything
wrong please open a jira, if you are at cluecon, find me and let me know
the bug number

K

On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net> wrote:

> **
> Where is the branch so I can checkout local?
>
> D:\fsnew>git branch -a
> * master
>   remotes/origin/FS-3432
>   remotes/origin/FS-4062
>   remotes/origin/HEAD -> origin/master
>   remotes/origin/dingaling_video
>   remotes/origin/master
>   remotes/origin/smgfs
>   remotes/origin/stable-test/freeswitch-1.2
>   remotes/origin/swk/fs_test
>   remotes/test/master
> --Dave
>
>  ------------------------------
> *From:* Ken Rice [mailto:krice at freeswitch.org]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org],
> freeswitch-dev at lists.freeswitch.org
> *Sent:* Tue, 07 Aug 2012 09:03:34 -0700
>
> *Subject:* [Freeswitch-users] Announcing FreeSWITCH 1.2
>
> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!
>
> Get your copy today at
> http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !
>
> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
> continue to fix bugs and security issues. giving you a stable platform for
> at least one year.
>
> Grab it today!
>
> The FreeSWITCH Team
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From drk at drkngs.net  Tue Aug  7 22:23:54 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Tue, 07 Aug 2012 11:23:54 -0700
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAPcNkdeC5Yp3ht_4KqYzrHEpVUAYuz_R9gjYi2EDmEepVqLFBQ@mail.gmail.com>
Message-ID: <20120807182354.68ae72ec@mail.tritonwest.net>

I also checked tags, but didn't see it. Are you sure your branch is tracked upstream?  
   
--Dave
      _____  

  From: Ken Rice [mailto:krice at freeswitch.org]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Tue, 07 Aug 2012 11:15:50 -0700
Subject: Re: [Freeswitch-users] Announcing FreeSWITCH 1.2

its Tagged, right now, we will be branching it shortly for tracking patches. Please note we had a couple of good users point out some smallish bugs so we're int he process of fixing those! so guys if you find anything wrong please open a jira, if you are at cluecon, find me and let me know the bug number  

  
K

  
On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net> wrote:
    
  
Where is the branch so I can checkout local?  
   
D:\fsnew>git branch -a
* master
  remotes/origin/FS-3432
  remotes/origin/FS-4062
  remotes/origin/HEAD -> origin/master
  remotes/origin/dingaling_video
  remotes/origin/master
  remotes/origin/smgfs
  remotes/origin/stable-test/freeswitch-1.2
  remotes/origin/swk/fs_test
  remotes/test/master
  
--Dave
      _____  

  From: Ken Rice [mailto:krice at freeswitch.org]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org], freeswitch-dev at lists.freeswitch.org
Sent: Tue, 07 Aug 2012 09:03:34 -0700  

Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2

  
  
The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!   

  
Get your copy today at http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !   

  
Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will continue to fix bugs and security issues. giving you a stable platform for at least one year.  

  
Grab it today!  

  
The FreeSWITCH Team   
   
 
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


      
   
 
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From krice at freeswitch.org  Tue Aug  7 22:31:49 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 7 Aug 2012 13:31:49 -0500
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <20120807182354.68ae72ec@mail.tritonwest.net>
References: <CAPcNkdeC5Yp3ht_4KqYzrHEpVUAYuz_R9gjYi2EDmEepVqLFBQ@mail.gmail.com>
	<20120807182354.68ae72ec@mail.tritonwest.net>
Message-ID: <CAPcNkddEP9TtOrLu=ikyN_B23QZZfOVBvyqA4Ogx+WDM2V0-uA@mail.gmail.com>

yes its there... git pull --tags

On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net> wrote:

> **
> I also checked tags, but didn't see it. Are you sure your branch is
> tracked upstream?
>
> --Dave
>
>  ------------------------------
> *From:* Ken Rice [mailto:krice at freeswitch.org]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> *Sent:* Tue, 07 Aug 2012 11:15:50 -0700
> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>
>
> its Tagged, right now, we will be branching it shortly for tracking
> patches. Please note we had a couple of good users point out some smallish
> bugs so we're int he process of fixing those! so guys if you find anything
> wrong please open a jira, if you are at cluecon, find me and let me know
> the bug number
>
> K
>
> On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>
>> **
>> Where is the branch so I can checkout local?
>>
>> D:\fsnew>git branch -a
>> * master
>>   remotes/origin/FS-3432
>>   remotes/origin/FS-4062
>>   remotes/origin/HEAD -> origin/master
>>   remotes/origin/dingaling_video
>>   remotes/origin/master
>>   remotes/origin/smgfs
>>   remotes/origin/stable-test/freeswitch-1.2
>>   remotes/origin/swk/fs_test
>>   remotes/test/master
>> --Dave
>>
>>  ------------------------------
>> *From:* Ken Rice [mailto:krice at freeswitch.org]
>> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org],
>> freeswitch-dev at lists.freeswitch.org
>> *Sent:* Tue, 07 Aug 2012 09:03:34 -0700
>>
>> *Subject:* [Freeswitch-users] Announcing FreeSWITCH 1.2
>>
>>  The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!
>>
>> Get your copy today at
>> http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !
>>
>> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
>> continue to fix bugs and security issues. giving you a stable platform for
>> at least one year.
>>
>> Grab it today!
>>
>> The FreeSWITCH Team
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From drk at drkngs.net  Tue Aug  7 22:43:41 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Tue, 07 Aug 2012 11:43:41 -0700
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAPcNkddEP9TtOrLu=ikyN_B23QZZfOVBvyqA4Ogx+WDM2V0-uA@mail.gmail.com>
Message-ID: <20120807184341.2369f896@mail.tritonwest.net>

Got it, and guess what. It don't compile on windows :P hehe  
   
--Dave
      _____  

  From: Ken Rice [mailto:krice at freeswitch.org]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Tue, 07 Aug 2012 11:31:49 -0700
Subject: Re: [Freeswitch-users] Announcing FreeSWITCH 1.2

yes its there... git pull --tags
  

  
On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net> wrote:
    
  
I also checked tags, but didn't see it. Are you sure your branch is tracked upstream?  
   
--Dave
    
    _____  

  From: Ken Rice [mailto:krice at freeswitch.org]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Tue, 07 Aug 2012 11:15:50 -0700
Subject: Re: [Freeswitch-users] Announcing FreeSWITCH 1.2  
  


its Tagged, right now, we will be branching it shortly for tracking patches. Please note we had a couple of good users point out some smallish bugs so we're int he process of fixing those! so guys if you find anything wrong please open a jira, if you are at cluecon, find me and let me know the bug number   

  
K

  
On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net> wrote:
    
  
Where is the branch so I can checkout local?  
   
D:\fsnew>git branch -a
* master
  remotes/origin/FS-3432
  remotes/origin/FS-4062
  remotes/origin/HEAD -> origin/master
  remotes/origin/dingaling_video
  remotes/origin/master
  remotes/origin/smgfs
  remotes/origin/stable-test/freeswitch-1.2
  remotes/origin/swk/fs_test
  remotes/test/master
  
--Dave
      _____  

  From: Ken Rice [mailto:krice at freeswitch.org]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org], freeswitch-dev at lists.freeswitch.org
Sent: Tue, 07 Aug 2012 09:03:34 -0700   

Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2

  
  
The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!   

  
Get your copy today at http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !   

  
Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will continue to fix bugs and security issues. giving you a stable platform for at least one year.  

  
Grab it today!  

  
The FreeSWITCH Team   
   
 
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


  
   
 
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


      
   
 
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From aksrini at hotmail.com  Tue Aug  7 23:22:37 2012
From: aksrini at hotmail.com (Srini K)
Date: Tue, 7 Aug 2012 12:22:37 -0700
Subject: [Freeswitch-users] Head version not compiling on windows
	(libspandsp issue)
Message-ID: <BAY170-W4644B714CD753AE88D0090AACE0@phx.gbl>





Hi,
Today I updated the FreeSWITCH to head version and tried to compile it on Windows(Visual Studio 2010). 
I get the folloing errors on libspandspError 9 error LNK2019: unresolved external symbol _set_lab_gamut at 32 referenced in function _read_tiff_image C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp
Error 10 error LNK2019: unresolved external symbol _set_lab_illuminant at 16 referenced in function _read_tiff_image C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp
Error 11 error LNK2019: unresolved external symbol _lab_to_srgb at 16 referenced in function _row_read C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp
Error 12 error LNK1120: 3 unresolved externals C:\FreeSWITCH\Win32\Debug\libspandsp.dll libspandsp Any ideas? Thanks
Srini 		 	   		  
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From miha at softnet.si  Tue Aug  7 23:22:49 2012
From: miha at softnet.si (Miha)
Date: Tue, 07 Aug 2012 21:22:49 +0200
Subject: [Freeswitch-users] upadte FS (make current)
In-Reply-To: <5020EC2A.2080302@softnet.si>
References: <1FFF97C269757C458224B7C895F35F1513F759@cantor.std.visionutv.se>
	<5020EC2A.2080302@softnet.si>
Message-ID: <web-86139963@softnet.si>

HI,

just one more information. What is the best way to see what
leaks memory and to be sure that FS does not leak it?

If I look at the top command I can see that the free memory
is slowly dropping, but from ps aux I can see that FS
consums 1.3% of memory (the same).

Thanks!

Miha

Thanks!
 MIha

On Tue, 07 Aug 2012 12:21:30 +0200
 Miha <miha at softnet.si> wrote:
> Hi Peter,
> 
> fsctl reclaim_mem this will not harm any call and etc?
> 
> So if freeswitch does not leak memory what should I see
> in top (memory will not drop?)?
> 
> Thanks!
> Miha
> 
> 
> 
> On 8/7/2012 11:48 AM, Peter Olsson wrote:
> > It's always hard to say what is a memory leak and
> what's not. FS uses memory pools internally, which
> sometime might make it look like it leaks memory, but in
> the end FS will reuse that memory. You can also run
> "fsctl reclaim_mem" in the console or fs_cli, and see if
> that makes the memory usage drop.
> >
> > /Peter
> >
> > -----Ursprungligt meddelande-----
> > Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
> F?r Miha
> > Skickat: den 7 augusti 2012 11:14
> > Till: FreeSWITCH Users Help
> > ?mne: Re: [Freeswitch-users] upadte FS (make current)
> >
> > On 8/7/2012 10:54 AM, Peter Olsson wrote:
> >> Usually it's no problem. However, I always recommend
> to have a test
> >> server to run basic tests before going into
> production. There have
> >> been a couple of memory leaks fixed, but I think that
> was before your
> >> current version - but I don't remember all the commits
> from my head..
> >> :)
> >>
> >> I would also suggest to read through the git commit
> log, to get an idea of what has been done.
> >>
> >> /Peter
> >>
> >>
> >> -----Ursprungligt meddelande-----
> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
> F?r Miha
> >> Skickat: den 7 augusti 2012 10:37
> >> Till: FreeSWITCH Users Help
> >> ?mne: [Freeswitch-users] upadte FS (make current)
> >>
> >> Hi,
> >>
> >> I am running on production server FS (runs ok). I
> would like to upgrade it to latest git.
> >> Can I expect any problems? I not using any special
> module (just mod_rad_auth, for cdr mod_xml_cdr).
> >>
> >> Is it better to first test everything on test server
> and than make upgrade?
> >>
> >> My FS is now up for 13 days. I am looking at memory
> for a few days and noticing that free memory is shrinking
> (slowly). Was there any problem with memory leak? :
> >>
> >> top - 09:46:46 up 13 days, 12:32,  3 users,  load
> average: 0.00, 0.04, 0.05
> >> Tasks: 159 total,   1 running, 158 sleeping,   0
> stopped,   0 zombie
> >> Cpu(s):  0.6%us,  0.4%sy,  0.0%ni, 99.0%id,  0.0%wa,
>  0.0%hi, 0.0%si, 0.0%st
> >> Mem:   4046636k total,  3356340k used,   690296k free,
>   357296k buffers
> >> Swap:  6094840k total,        0k used,  6094840k free,
>  2629280k cached
> >>
> >> My version: FreeSWITCH Version
> >> 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5
> >> (1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul
> 2012 15:18:27 Z)
> >>
> >> Thanks!
> >>
> >> Miha
> >>
> >>
>
______________________________________________________________________
> >> ___ Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >>
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> rs
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >>
>
______________________________________________________________________
> >> ___ Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >>
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> rs
> >> http://www.freeswitch.org
> >>
> > @Peter thanks for your quick answer:)
> >
> > I readed jira postes and noticed that this was befor my
> git version (as you said).
> > If there was leak problem (I do not know if it is, just
> noticing that free memory is droping), would FS be stable
> for 13 days or would this bug be present sooner (memory
> leak)?
> >
> > regards,
> > Miha
> >
> >
> >
>
_________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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:5020daa032762709953773!
> >
> >
> >
>
_________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> >
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
>
_________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Tue Aug  7 23:51:02 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 7 Aug 2012 19:51:02 +0000
Subject: [Freeswitch-users] upadte FS (make current)
In-Reply-To: <web-86139963@softnet.si>
References: <1FFF97C269757C458224B7C895F35F1513F759@cantor.std.visionutv.se>
	<5020EC2A.2080302@softnet.si>,<web-86139963@softnet.si>
Message-ID: <1FFF97C269757C458224B7C895F35F1513FFDD@cantor.std.visionutv.se>

Why do you even think FS leaks memory? How much memory does it use?

Anyway, to look for memory leaks, check out this: 
http://wiki.freeswitch.org/wiki/Reporting_Bugs#Collection_Information_With_Valgrind_.28Linux.2FUnix.29

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Miha [miha at softnet.si]
Skickat: den 7 augusti 2012 21:22
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] upadte FS (make current)

HI,

just one more information. What is the best way to see what
leaks memory and to be sure that FS does not leak it?

If I look at the top command I can see that the free memory
is slowly dropping, but from ps aux I can see that FS
consums 1.3% of memory (the same).

Thanks!

Miha

Thanks!
 MIha

On Tue, 07 Aug 2012 12:21:30 +0200
 Miha <miha at softnet.si> wrote:
> Hi Peter,
>
> fsctl reclaim_mem this will not harm any call and etc?
>
> So if freeswitch does not leak memory what should I see
> in top (memory will not drop?)?
>
> Thanks!
> Miha
>
>
>
> On 8/7/2012 11:48 AM, Peter Olsson wrote:
> > It's always hard to say what is a memory leak and
> what's not. FS uses memory pools internally, which
> sometime might make it look like it leaks memory, but in
> the end FS will reuse that memory. You can also run
> "fsctl reclaim_mem" in the console or fs_cli, and see if
> that makes the memory usage drop.
> >
> > /Peter
> >
> > -----Ursprungligt meddelande-----
> > Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
> F?r Miha
> > Skickat: den 7 augusti 2012 11:14
> > Till: FreeSWITCH Users Help
> > ?mne: Re: [Freeswitch-users] upadte FS (make current)
> >
> > On 8/7/2012 10:54 AM, Peter Olsson wrote:
> >> Usually it's no problem. However, I always recommend
> to have a test
> >> server to run basic tests before going into
> production. There have
> >> been a couple of memory leaks fixed, but I think that
> was before your
> >> current version - but I don't remember all the commits
> from my head..
> >> :)
> >>
> >> I would also suggest to read through the git commit
> log, to get an idea of what has been done.
> >>
> >> /Peter
> >>
> >>
> >> -----Ursprungligt meddelande-----
> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
> F?r Miha
> >> Skickat: den 7 augusti 2012 10:37
> >> Till: FreeSWITCH Users Help
> >> ?mne: [Freeswitch-users] upadte FS (make current)
> >>
> >> Hi,
> >>
> >> I am running on production server FS (runs ok). I
> would like to upgrade it to latest git.
> >> Can I expect any problems? I not using any special
> module (just mod_rad_auth, for cdr mod_xml_cdr).
> >>
> >> Is it better to first test everything on test server
> and than make upgrade?
> >>
> >> My FS is now up for 13 days. I am looking at memory
> for a few days and noticing that free memory is shrinking
> (slowly). Was there any problem with memory leak? :
> >>
> >> top - 09:46:46 up 13 days, 12:32,  3 users,  load
> average: 0.00, 0.04, 0.05
> >> Tasks: 159 total,   1 running, 158 sleeping,   0
> stopped,   0 zombie
> >> Cpu(s):  0.6%us,  0.4%sy,  0.0%ni, 99.0%id,  0.0%wa,
>  0.0%hi, 0.0%si, 0.0%st
> >> Mem:   4046636k total,  3356340k used,   690296k free,
>   357296k buffers
> >> Swap:  6094840k total,        0k used,  6094840k free,
>  2629280k cached
> >>
> >> My version: FreeSWITCH Version
> >> 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5
> >> (1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul
> 2012 15:18:27 Z)
> >>
> >> Thanks!
> >>
> >> Miha
> >>
> >>
>
______________________________________________________________________
> >> ___ Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >>
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> rs
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >>
>
______________________________________________________________________
> >> ___ Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >>
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> rs
> >> http://www.freeswitch.org
> >>
> > @Peter thanks for your quick answer:)
> >
> > I readed jira postes and noticed that this was befor my
> git version (as you said).
> > If there was leak problem (I do not know if it is, just
> noticing that free memory is droping), would FS be stable
> for 13 days or would this bug be present sooner (memory
> leak)?
> >
> > regards,
> > Miha
> >
> >
> >
>
_________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> >
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> >
>
_________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> >
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
_________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at 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:5021693632761271914539!



From peter.olsson at visionutveckling.se  Tue Aug  7 23:51:43 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 7 Aug 2012 19:51:43 +0000
Subject: [Freeswitch-users] Head version not compiling on
	windows	(libspandsp issue)
In-Reply-To: <BAY170-W4644B714CD753AE88D0090AACE0@phx.gbl>
References: <BAY170-W4644B714CD753AE88D0090AACE0@phx.gbl>
Message-ID: <1FFF97C269757C458224B7C895F35F1513FFE7@cantor.std.visionutv.se>

Jira please

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Srini K [aksrini at hotmail.com]
Skickat: den 7 augusti 2012 21:22
Till: fs
?mne: [Freeswitch-users] Head version not compiling on windows (libspandsp issue)

Hi,
Today I updated the FreeSWITCH to head version and tried to compile it on Windows(Visual Studio 2010).
I get the folloing errors on libspandsp
Error 9 error LNK2019: unresolved external symbol _set_lab_gamut at 32<mailto:_set_lab_gamut at 32> referenced in function _read_tiff_image C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp
Error 10 error LNK2019: unresolved external symbol _set_lab_illuminant at 16<mailto:_set_lab_illuminant at 16> referenced in function _read_tiff_image C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp
Error 11 error LNK2019: unresolved external symbol _lab_to_srgb at 16<mailto:_lab_to_srgb at 16> referenced in function _row_read C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp
Error 12 error LNK1120: 3 unresolved externals C:\FreeSWITCH\Win32\Debug\libspandsp.dll libspandsp

Any ideas?

Thanks
Srini
!DSPAM:5021692d32762117992842!


From toddb at toddbailey.net  Tue Aug  7 23:53:48 2012
From: toddb at toddbailey.net (toddbailey)
Date: Tue, 7 Aug 2012 12:53:48 -0700 (PDT)
Subject: [Freeswitch-users] Building Skype Issues
Message-ID: <1344369228639-7581628.post@n2.nabble.com>

Hi All,

After a dreadful several weeks of trying to get my Fedora 17 upgrade from
F14 to cooperate, I'm now running Mint 13 LST x64 as my primary server.

Now that most of the drama has past, I'm back to trying to get mod_skypopen
to install.
Before F17 I was running F14 so you may see why the upgrade was required.
'nough said,

Just like F14 I'm seeing an error in running the install sequence

cd /usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss
make clean; make; insmod ./skypopen.ko; mknod /dev/dsp c 14 3

once again I see  
insmod: error inserting './skypopen.ko': -1 Device or resource busy
mknod: `/dev/dsp': File exists

I'm told previously that I don't want to see any snd modules listed in the
command

lsmod |grep snd


but in my case I see several

lsmod | grep snd

snd_hda_codec_realtek   223962  1 
snd_hda_intel          33773  3 
snd_hda_codec         127706  2 snd_hda_codec_realtek,snd_hda_intel
snd_hwdep              13668  1 snd_hda_codec
snd_pcm                97188  2 snd_hda_intel,snd_hda_codec
snd_seq_midi           13324  0 
snd_rawmidi            30748  1 snd_seq_midi
snd_seq_midi_event     14899  1 snd_seq_midi
snd_seq                61896  2 snd_seq_midi,snd_seq_midi_event
snd_timer              29990  2 snd_pcm,snd_seq
snd_seq_device         14540  3 snd_seq_midi,snd_rawmidi,snd_seq
snd                    78855  15
snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device
soundcore              15091  1 snd
snd_page_alloc         18529  2 snd_hda_intel,snd_pcm

Questions:

do I need to rmmod each of these before I can successfully install the
mod_skypopen ?

Assuming this is true,  can I insmod the list to return to the previous
state or does the install of mod_skypopen limit what I can do with the
systems audio capabilities?
I utilize other applications that require audio ie: alsa and pulse modules
to function.

Thanks



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gmaruzz at gmail.com  Wed Aug  8 00:04:38 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 7 Aug 2012 22:04:38 +0200
Subject: [Freeswitch-users] Building Skype Issues
In-Reply-To: <1344369228639-7581628.post@n2.nabble.com>
References: <1344369228639-7581628.post@n2.nabble.com>
Message-ID: <CALXCt0ojh_AvsBLhDiaUHdq8PU-_fJtEXaoF9cfyC2JtASwh3Q@mail.gmail.com>

you better follow the wikipage, that tells you to use centos 6.x or ubuntu,
server installations, at 64 bit, or windows.

If you choose to use a different OS, you need to know how to deal with it.

1) do not use a desktop distro, it will not work
2) yes, rmmod all the snd* (but, why are there? You want them not loaded at
all. Uloading is a hack)

so, to avoid frustration, use a supported OS.

In all other cases, you are going to use an awful amount of time and get
lot of frustrations, I can guarantee you ;).

-giovanni



On Tue, Aug 7, 2012 at 9:53 PM, toddbailey <toddb at toddbailey.net> wrote:

> Hi All,
>
> After a dreadful several weeks of trying to get my Fedora 17 upgrade from
> F14 to cooperate, I'm now running Mint 13 LST x64 as my primary server.
>
> Now that most of the drama has past, I'm back to trying to get mod_skypopen
> to install.
> Before F17 I was running F14 so you may see why the upgrade was required.
> 'nough said,
>
> Just like F14 I'm seeing an error in running the install sequence
>
> cd /usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss
> make clean; make; insmod ./skypopen.ko; mknod /dev/dsp c 14 3
>
> once again I see
> insmod: error inserting './skypopen.ko': -1 Device or resource busy
> mknod: `/dev/dsp': File exists
>
> I'm told previously that I don't want to see any snd modules listed in the
> command
>
> lsmod |grep snd
>
>
> but in my case I see several
>
> lsmod | grep snd
>
> snd_hda_codec_realtek   223962  1
> snd_hda_intel          33773  3
> snd_hda_codec         127706  2 snd_hda_codec_realtek,snd_hda_intel
> snd_hwdep              13668  1 snd_hda_codec
> snd_pcm                97188  2 snd_hda_intel,snd_hda_codec
> snd_seq_midi           13324  0
> snd_rawmidi            30748  1 snd_seq_midi
> snd_seq_midi_event     14899  1 snd_seq_midi
> snd_seq                61896  2 snd_seq_midi,snd_seq_midi_event
> snd_timer              29990  2 snd_pcm,snd_seq
> snd_seq_device         14540  3 snd_seq_midi,snd_rawmidi,snd_seq
> snd                    78855  15
>
> snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device
> soundcore              15091  1 snd
> snd_page_alloc         18529  2 snd_hda_intel,snd_pcm
>
> Questions:
>
> do I need to rmmod each of these before I can successfully install the
> mod_skypopen ?
>
> Assuming this is true,  can I insmod the list to return to the previous
> state or does the install of mod_skypopen limit what I can do with the
> systems audio capabilities?
> I utilize other applications that require audio ie: alsa and pulse modules
> to function.
>
> Thanks
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 drk at drkngs.net  Wed Aug  8 00:06:03 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Tue, 07 Aug 2012 13:06:03 -0700
Subject: [Freeswitch-users]
	=?iso-8859-1?q?Head_version_not_compiling_on?=
	=?iso-8859-1?q?=09windows=09=28libspandsp_issue=29?=
In-Reply-To: <1FFF97C269757C458224B7C895F35F1513FFE7@cantor.std.visionutv.se>
Message-ID: <20120807200603.942e78de@mail.tritonwest.net>

It looks a little worse. From what I can tell SPANDSP now uses libraries that arn't there on windows, and I can't find them.
      _____  

  From: Peter Olsson [mailto:peter.olsson at visionutveckling.se]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Tue, 07 Aug 2012 12:51:43 -0700
Subject: Re: [Freeswitch-users] Head version not compiling on windows (libspandsp issue)

Jira please

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Srini K [aksrini at hotmail.com]
Skickat: den 7 augusti 2012 21:22
Till: fs
?mne: [Freeswitch-users] Head version not compiling on windows (libspandsp issue)

Hi,
Today I updated the FreeSWITCH to head version and tried to compile it on Windows(Visual Studio 2010).
I get the folloing errors on libspandsp
Error 9 error LNK2019: unresolved external symbol _set_lab_gamut at 32<mailto:_set_lab_gamut at 32> referenced in function _read_tiff_image C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp
Error 10 error LNK2019: unresolved external symbol _set_lab_illuminant at 16<mailto:_set_lab_illuminant at 16> referenced in function _read_tiff_image C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp
Error 11 error LNK2019: unresolved external symbol _lab_to_srgb at 16<mailto:_lab_to_srgb at 16> referenced in function _row_read C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp
Error 12 error LNK1120: 3 unresolved externals C:\FreeSWITCH\Win32\Debug\libspandsp.dll libspandsp

Any ideas?

Thanks
Srini
!DSPAM:5021692d32762117992842!

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
      
   
 
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From krice at freeswitch.org  Wed Aug  8 00:18:25 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 7 Aug 2012 15:18:25 -0500
Subject: [Freeswitch-users] Head version not compiling on windows
 (libspandsp issue)
In-Reply-To: <20120807200603.942e78de@mail.tritonwest.net>
References: <1FFF97C269757C458224B7C895F35F1513FFE7@cantor.std.visionutv.se>
	<20120807200603.942e78de@mail.tritonwest.net>
Message-ID: <CAPcNkdcokJT++RCdWj3hbFCMJ=PGTpwW9eTAEqkSwrvUtp=P4Q@mail.gmail.com>

Steve made a few changes this morning... please open a JIRA so we can track
this and get it fixed once he gets back around

On Tue, Aug 7, 2012 at 3:06 PM, Dave R. Kompel <drk at drkngs.net> wrote:

> **
> It looks a little worse. From what I can tell SPANDSP now uses libraries
> that arn't there on windows, and I can't find them.
>
>  ------------------------------
> *From:* Peter Olsson [mailto:peter.olsson at visionutveckling.se]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> *Sent:* Tue, 07 Aug 2012 12:51:43 -0700
> *Subject:* Re: [Freeswitch-users] Head version not compiling on windows
> (libspandsp issue)
>
>
> Jira please
>
> /Peter
> ________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f?r Srini K [
> aksrini at hotmail.com]
> Skickat: den 7 augusti 2012 21:22
> Till: fs
> ?mne: [Freeswitch-users] Head version not compiling on windows (libspandsp
> issue)
>
> Hi,
> Today I updated the FreeSWITCH to head version and tried to compile it on
> Windows(Visual Studio 2010).
> I get the folloing errors on libspandsp
> Error 9 error LNK2019: unresolved external symbol _set_lab_gamut at 32
> <mailto:_set_lab_gamut at 32> referenced in function _read_tiff_image
> C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp
> Error 10 error LNK2019: unresolved external symbol _set_lab_illuminant at 16
> <mailto:_set_lab_illuminant at 16> referenced in function _read_tiff_image
> C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp
> Error 11 error LNK2019: unresolved external symbol _lab_to_srgb at 16<mailto:
> _lab_to_srgb at 16> referenced in function _row_read
> C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp
> Error 12 error LNK1120: 3 unresolved externals
> C:\FreeSWITCH\Win32\Debug\libspandsp.dll libspandsp
>
> Any ideas?
>
> Thanks
> Srini
> !DSPAM:5021692d32762117992842!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From toddb at toddbailey.net  Wed Aug  8 00:19:47 2012
From: toddb at toddbailey.net (toddbailey)
Date: Tue, 7 Aug 2012 13:19:47 -0700 (PDT)
Subject: [Freeswitch-users] Building Skype Issues
In-Reply-To: <CALXCt0ojh_AvsBLhDiaUHdq8PU-_fJtEXaoF9cfyC2JtASwh3Q@mail.gmail.com>
References: <1344369228639-7581628.post@n2.nabble.com>
	<CALXCt0ojh_AvsBLhDiaUHdq8PU-_fJtEXaoF9cfyC2JtASwh3Q@mail.gmail.com>
Message-ID: <1344370768.5586.7.camel@mythtv>


Mint 13 LST is a long term supported product based off Ubuntu 12 so it
should supported at least for the most part.  Any minor nuances I can
deal with.

What exactly do you mean do not use a desktop distro ?
	I need the gnome desktop for other activities.
	
yes, rmmod all the snd* (but, why are there?)
Because other applications installed use sound modules.

This is a multipurpose server that provides many different functions, a
pbx is just one of several functions.

Should I consider building a virtual machine and running free switch in
it instead on the core os?


On Tue, 2012-08-07 at 13:08 -0700, Giovanni Maruzzelli-2 [via
freeswitch-users] wrote:
> you better follow the wikipage, that tells you to use centos 6.x or
> ubuntu, server installations, at 64 bit, or windows.
> 
> If you choose to use a different OS, you need to know how to deal with
> it.
> 
> 1) do not use a desktop distro, it will not work
> 2) yes, rmmod all the snd* (but, why are there? You want them not
> loaded at all. Uloading is a hack)
> 
> so, to avoid frustration, use a supported OS.
> 
> In all other cases, you are going to use an awful amount of time and
> get lot of frustrations, I can guarantee you ;).
> 
> -giovanni
> 
> 
> 
> On Tue, Aug 7, 2012 at 9:53 PM, toddbailey <[hidden email]> wrote:
>         Hi All,
>         
>         After a dreadful several weeks of trying to get my Fedora 17
>         upgrade from
>         F14 to cooperate, I'm now running Mint 13 LST x64 as my
>         primary server.
>         
>         Now that most of the drama has past, I'm back to trying to get
>         mod_skypopen
>         to install.
>         Before F17 I was running F14 so you may see why the upgrade
>         was required.
>         'nough said,
>         
>         Just like F14 I'm seeing an error in running the install
>         sequence
>         
>         cd /usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss
>         make clean; make; insmod ./skypopen.ko; mknod /dev/dsp c 14 3
>         
>         once again I see
>         insmod: error inserting './skypopen.ko': -1 Device or resource
>         busy
>         mknod: `/dev/dsp': File exists
>         
>         I'm told previously that I don't want to see any snd modules
>         listed in the
>         command
>         
>         lsmod |grep snd
>         
>         
>         but in my case I see several
>         
>         lsmod | grep snd
>         
>         snd_hda_codec_realtek   223962  1
>         snd_hda_intel          33773  3
>         snd_hda_codec         127706  2
>         snd_hda_codec_realtek,snd_hda_intel
>         snd_hwdep              13668  1 snd_hda_codec
>         snd_pcm                97188  2 snd_hda_intel,snd_hda_codec
>         snd_seq_midi           13324  0
>         snd_rawmidi            30748  1 snd_seq_midi
>         snd_seq_midi_event     14899  1 snd_seq_midi
>         snd_seq                61896  2
>         snd_seq_midi,snd_seq_midi_event
>         snd_timer              29990  2 snd_pcm,snd_seq
>         snd_seq_device         14540  3
>         snd_seq_midi,snd_rawmidi,snd_seq
>         snd                    78855  15
>         snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device
>         soundcore              15091  1 snd
>         snd_page_alloc         18529  2 snd_hda_intel,snd_pcm
>         
>         Questions:
>         
>         do I need to rmmod each of these before I can successfully
>         install the
>         mod_skypopen ?
>         
>         Assuming this is true,  can I insmod the list to return to the
>         previous
>         state or does the install of mod_skypopen limit what I can do
>         with the
>         systems audio capabilities?
>         I utilize other applications that require audio ie: alsa and
>         pulse modules
>         to function.
>         
>         Thanks
>         
>         
>         
>         --
>         View this message in context:
>         http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628.html
>         Sent from the freeswitch-users mailing list archive at
>         Nabble.com.
>         
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         [hidden email]
>         http://www.freeswitchsolutions.com
>         
>         
>         
>         
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>         
>         Join Us At ClueCon - Aug 7-9, 2012
>         
>         FreeSWITCH-users mailing list
>         [hidden email]
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
> 
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> 
> 
> _________________________________________________________________________ 
> Professional FreeSWITCH Consulting Services: 
> [hidden email] 
> http://www.freeswitchsolutions.com
> 
>  
> 
> 
> Official FreeSWITCH Sites 
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012 
> 
> FreeSWITCH-users mailing list 
> [hidden email] 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> ______________________________________________________________________
> If you reply to this email, your message will be added to the
> discussion below:
> http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628p7581631.html 
> To unsubscribe from Building Skype Issues, click here.
> NAML 

-- 
-
-
-    Best Regards,
-
-            Todd Bailey
-
-





--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628p7581633.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From aksrini at hotmail.com  Wed Aug  8 00:21:51 2012
From: aksrini at hotmail.com (Srini K)
Date: Tue, 7 Aug 2012 13:21:51 -0700
Subject: [Freeswitch-users] Head version not compiling on windows
 (libspandsp issue)
In-Reply-To: <CAPcNkdcokJT++RCdWj3hbFCMJ=PGTpwW9eTAEqkSwrvUtp=P4Q@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F1513FFE7@cantor.std.visionutv.se>,
	<20120807200603.942e78de@mail.tritonwest.net>,
	<CAPcNkdcokJT++RCdWj3hbFCMJ=PGTpwW9eTAEqkSwrvUtp=P4Q@mail.gmail.com>
Message-ID: <BAY170-W12495B57EC8FA819EDB7B5CAACE0@phx.gbl>


http://jira.freeswitch.org/browse/FS-4504
 Date: Tue, 7 Aug 2012 15:18:25 -0500
From: krice at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Head version not compiling on windows (libspandsp issue)

Steve made a few changes this morning... please open a JIRA so we can track this and get it fixed once he gets back around

On Tue, Aug 7, 2012 at 3:06 PM, Dave R. Kompel <drk at drkngs.net> wrote:





It looks a little worse. From what I can tell SPANDSP now uses libraries that arn't there on windows, and I can't find them.



From: Peter Olsson [mailto:peter.olsson at visionutveckling.se]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]


Sent: Tue, 07 Aug 2012 12:51:43 -0700
Subject: Re: [Freeswitch-users] Head version not compiling on windows (libspandsp issue)

Jira please

/Peter
________________________________


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Srini K [aksrini at hotmail.com]


Skickat: den 7 augusti 2012 21:22
Till: fs
?mne: [Freeswitch-users] Head version not compiling on windows (libspandsp issue)

Hi,
Today I updated the FreeSWITCH to head version and tried to compile it on Windows(Visual Studio 2010).


I get the folloing errors on libspandsp
Error 9 error LNK2019: unresolved external symbol _set_lab_gamut at 32<mailto:_set_lab_gamut at 32> referenced in function _read_tiff_image C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp


Error 10 error LNK2019: unresolved external symbol _set_lab_illuminant at 16<mailto:_set_lab_illuminant at 16> referenced in function _read_tiff_image C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp


Error 11 error LNK2019: unresolved external symbol _lab_to_srgb at 16<mailto:_lab_to_srgb at 16> referenced in function _row_read C:\FreeSWITCH\libs\spandsp\src\t4_tx.obj libspandsp


Error 12 error LNK1120: 3 unresolved externals C:\FreeSWITCH\Win32\Debug\libspandsp.dll libspandsp

Any ideas?

Thanks
Srini
!DSPAM:5021692d32762117992842!

_________________________________________________________________________


Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com






Official FreeSWITCH Sites
http://www.freeswitch.org


http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list


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


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





 
 
_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com



Join Us At ClueCon - Aug 7-9, 2012



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

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

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

http://www.freeswitch.org





_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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  Wed Aug  8 00:33:24 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 7 Aug 2012 22:33:24 +0200
Subject: [Freeswitch-users] Building Skype Issues
In-Reply-To: <1344370768.5586.7.camel@mythtv>
References: <1344369228639-7581628.post@n2.nabble.com>
	<CALXCt0ojh_AvsBLhDiaUHdq8PU-_fJtEXaoF9cfyC2JtASwh3Q@mail.gmail.com>
	<1344370768.5586.7.camel@mythtv>
Message-ID: <CALXCt0o0b1YjjWe2+jFzitANkU=6fpKbEL4K9k=TgB1uiau-uw@mail.gmail.com>

On Tue, Aug 7, 2012 at 10:19 PM, toddbailey <toddb at toddbailey.net> wrote:

>
> Mint 13 LST is a long term supported product based off Ubuntu 12 so it
> should supported at least for the most part.  Any minor nuances I can
> deal with.
>
> What exactly do you mean do not use a desktop distro ?
>         I need the gnome desktop for other activities.
>


a desktop will not work with skypopen, skypopen is designed to run on a
server distro.

If you want to explore, try to understand what is starting the oss (open
sound system) alsa emulation on your system. Sorry, I don't know which
modules you'll have to unload. You probably need to modify the settings of
pulseaudio too.

Ask in alsa and puleaudio forums how you can unload the OSS emulation.

I repeat, you'll end up trashing a lot of time and be very frustrated.

I will no more step up in this, use a server install - I told you twice ;).
(I'm the developer of skypopen)

-giovanni






>
> yes, rmmod all the snd* (but, why are there?)
> Because other applications installed use sound modules.
>
> This is a multipurpose server that provides many different functions, a
> pbx is just one of several functions.
>
> Should I consider building a virtual machine and running free switch in
> it instead on the core os?
>
>
> On Tue, 2012-08-07 at 13:08 -0700, Giovanni Maruzzelli-2 [via
> freeswitch-users] wrote:
>
> > you better follow the wikipage, that tells you to use centos 6.x or
> > ubuntu, server installations, at 64 bit, or windows.
> >
> > If you choose to use a different OS, you need to know how to deal with
> > it.
> >
> > 1) do not use a desktop distro, it will not work
> > 2) yes, rmmod all the snd* (but, why are there? You want them not
> > loaded at all. Uloading is a hack)
> >
> > so, to avoid frustration, use a supported OS.
> >
> > In all other cases, you are going to use an awful amount of time and
> > get lot of frustrations, I can guarantee you ;).
> >
> > -giovanni
> >
> >
> >
> > On Tue, Aug 7, 2012 at 9:53 PM, toddbailey <[hidden email]> wrote:
> >         Hi All,
> >
> >         After a dreadful several weeks of trying to get my Fedora 17
> >         upgrade from
> >         F14 to cooperate, I'm now running Mint 13 LST x64 as my
> >         primary server.
> >
> >         Now that most of the drama has past, I'm back to trying to get
> >         mod_skypopen
> >         to install.
> >         Before F17 I was running F14 so you may see why the upgrade
> >         was required.
> >         'nough said,
> >
> >         Just like F14 I'm seeing an error in running the install
> >         sequence
> >
> >         cd /usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss
> >         make clean; make; insmod ./skypopen.ko; mknod /dev/dsp c 14 3
> >
> >         once again I see
> >         insmod: error inserting './skypopen.ko': -1 Device or resource
> >         busy
> >         mknod: `/dev/dsp': File exists
> >
> >         I'm told previously that I don't want to see any snd modules
> >         listed in the
> >         command
> >
> >         lsmod |grep snd
> >
> >
> >         but in my case I see several
> >
> >         lsmod | grep snd
> >
> >         snd_hda_codec_realtek   223962  1
> >         snd_hda_intel          33773  3
> >         snd_hda_codec         127706  2
> >         snd_hda_codec_realtek,snd_hda_intel
> >         snd_hwdep              13668  1 snd_hda_codec
> >         snd_pcm                97188  2 snd_hda_intel,snd_hda_codec
> >         snd_seq_midi           13324  0
> >         snd_rawmidi            30748  1 snd_seq_midi
> >         snd_seq_midi_event     14899  1 snd_seq_midi
> >         snd_seq                61896  2
> >         snd_seq_midi,snd_seq_midi_event
> >         snd_timer              29990  2 snd_pcm,snd_seq
> >         snd_seq_device         14540  3
> >         snd_seq_midi,snd_rawmidi,snd_seq
> >         snd                    78855  15
> >
> snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device
>
> >         soundcore              15091  1 snd
> >         snd_page_alloc         18529  2 snd_hda_intel,snd_pcm
> >
> >         Questions:
> >
> >         do I need to rmmod each of these before I can successfully
> >         install the
> >         mod_skypopen ?
> >
> >         Assuming this is true,  can I insmod the list to return to the
> >         previous
> >         state or does the install of mod_skypopen limit what I can do
> >         with the
> >         systems audio capabilities?
> >         I utilize other applications that require audio ie: alsa and
> >         pulse modules
> >         to function.
> >
> >         Thanks
> >
> >
> >
> >         --
> >         View this message in context:
> >
> http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628.html
> >         Sent from the freeswitch-users mailing list archive at
> >         Nabble.com.
> >
> >
> _________________________________________________________________________
> >         Professional FreeSWITCH Consulting Services:
> >         [hidden email]
> >         http://www.freeswitchsolutions.com
> >
> >         
> >         
> >
> >         Official FreeSWITCH Sites
> >         http://www.freeswitch.org
> >         http://wiki.freeswitch.org
> >         http://www.cluecon.com
> >
> >         Join Us At ClueCon - Aug 7-9, 2012
> >
> >         FreeSWITCH-users mailing list
> >         [hidden email]
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >
> >
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > Cell : +39-347-2665618
> >
> >
> >
> _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > [hidden email]
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > [hidden email]
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > ______________________________________________________________________
> > If you reply to this email, your message will be added to the
> > discussion below:
> >
> http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628p7581631.html
>
> > To unsubscribe from Building Skype Issues, click here.
> > NAML
>
> --
> -
> -
> -    Best Regards,
> -
> -            Todd Bailey
> -
> -
>
>
> ------------------------------
> View this message in context: Re: Building Skype Issues<http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628p7581633.html>
>
> Sent from the freeswitch-users mailing list archive<http://freeswitch-users.2379917.n2.nabble.com/>at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ben at langfeld.co.uk  Wed Aug  8 00:34:26 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Tue, 7 Aug 2012 21:34:26 +0100
Subject: [Freeswitch-users] Adhearsion <3 FreeSWITCH
Message-ID: <CAAyX+KEgjQ2=ap7aiU_pQegZ1=vzZ7uWmWqPLDPzOkOuq4tmUQ@mail.gmail.com>

One of the most often-heard questions from the Adhearsion community has
long been: "Do you guys support FreeSWITCH?". I joined the Adhearsion
Project with a great desire for it to support FreeSWITCH, and since then
we've been working towards making that happen. At the last
AdhearsionConf<http://adhearsion.com/conference>we announced the
future that was to become Adhearsion 2. A large part of
that vision was the ability to write your voice application once, and then
to enjoy the portability across any supported telephony engine. With the
release of Adhearsion 2.0 back in April, we made a huge leap in that
direction, adding first-class support for Voxeo
PRISM<http://voxeolabs.com/prism>.
Today, we are very happy to finally announce initial support for FreeSWITCH
in Adhearsion.

I would also like to take a moment to share the philosophy behind the
version numbers used by the Adhearsion team. We are big believers in
SemVer<http://semver.org/>.
To put it briefly: we accept the API promise and hold it sacred.
Application developers should always feel confident upgrading versions of
Adhearsion within a major number (eg. 2.0 to 2.1; even 2.0 to 2.9). We
promise to make sure that Adhearsion remains perfectly backward compatible
as long as the major version number remains the same. Anything else is a
bug and we promise to fix it.

So today marks the release of Adhearsion 2.1.0. Take a moment and see some
of the other exciting improvements we have made in addition to FreeSWITCH
support by reading the
Changelog<https://github.com/adhearsion/adhearsion/blob/v2.1.0/CHANGELOG.md>.
As always, we appreciate bug reports on Github
Issues<https://github.com/adhearsion/adhearsion/issues>and will be
happy to answer questions or provide assistance on the Adhearsion
community mailing list <http://groups.google.com/group/adhearsion>.

Regards,
Ben Langfeld
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From toddb at toddbailey.net  Wed Aug  8 00:59:28 2012
From: toddb at toddbailey.net (toddbailey)
Date: Tue, 7 Aug 2012 13:59:28 -0700 (PDT)
Subject: [Freeswitch-users] Building Skype Issues
In-Reply-To: <CALXCt0o0b1YjjWe2+jFzitANkU=6fpKbEL4K9k=TgB1uiau-uw@mail.gmail.com>
References: <1344369228639-7581628.post@n2.nabble.com>
	<CALXCt0ojh_AvsBLhDiaUHdq8PU-_fJtEXaoF9cfyC2JtASwh3Q@mail.gmail.com>
	<1344370768.5586.7.camel@mythtv>
	<CALXCt0o0b1YjjWe2+jFzitANkU=6fpKbEL4K9k=TgB1uiau-uw@mail.gmail.com>
Message-ID: <1344373154.5586.10.camel@mythtv>

Sounds like skypopen needs to be expanded to play nice with others.

Will creating a virtual machine and running a server command line only
version of a "supported" os  resolve the snd module issues?


On Tue, 2012-08-07 at 13:37 -0700, Giovanni Maruzzelli-2 [via
freeswitch-users] wrote:
> 
> 
> On Tue, Aug 7, 2012 at 10:19 PM, toddbailey <[hidden email]> wrote:
>         
>         Mint 13 LST is a long term supported product based off Ubuntu
>         12 so it 
>         should supported at least for the most part.  Any minor
>         nuances I can 
>         deal with. 
>         
>         What exactly do you mean do not use a desktop distro ? 
>                 I need the gnome desktop for other activities. 
> 
> 
> a desktop will not work with skypopen, skypopen is designed to run on
> a server distro.
> 
> If you want to explore, try to understand what is starting the oss
> (open sound system) alsa emulation on your system. Sorry, I don't know
> which modules you'll have to unload. You probably need to modify the
> settings of pulseaudio too.
> 
> Ask in alsa and puleaudio forums how you can unload the OSS emulation.
> 
> I repeat, you'll end up trashing a lot of time and be very frustrated.
> 
> I will no more step up in this, use a server install - I told you
> twice ;). (I'm the developer of skypopen)
> 
> -giovanni
> 
> 
> 
> 
>  
>                 
>         yes, rmmod all the snd* (but, why are there?) 
>         Because other applications installed use sound modules. 
>         
>         This is a multipurpose server that provides many different
>         functions, a 
>         pbx is just one of several functions. 
>         
>         Should I consider building a virtual machine and running free
>         switch in 
>         it instead on the core os? 
>         
>         
>         On Tue, 2012-08-07 at 13:08 -0700, Giovanni Maruzzelli-2 [via 
>         freeswitch-users] wrote: 
>         
>         > you better follow the wikipage, that tells you to use centos
>         6.x or 
>         > ubuntu, server installations, at 64 bit, or windows. 
>         > 
>         > If you choose to use a different OS, you need to know how to
>         deal with 
>         > it. 
>         > 
>         > 1) do not use a desktop distro, it will not work 
>         > 2) yes, rmmod all the snd* (but, why are there? You want
>         them not 
>         > loaded at all. Uloading is a hack) 
>         > 
>         > so, to avoid frustration, use a supported OS. 
>         > 
>         > In all other cases, you are going to use an awful amount of
>         time and 
>         > get lot of frustrations, I can guarantee you ;). 
>         > 
>         > -giovanni 
>         > 
>         > 
>         > 
>         > On Tue, Aug 7, 2012 at 9:53 PM, toddbailey <[hidden email]>
>         wrote: 
>         
>         >         Hi All, 
>         >         
>         >         After a dreadful several weeks of trying to get my
>         Fedora 17 
>         >         upgrade from 
>         >         F14 to cooperate, I'm now running Mint 13 LST x64 as
>         my 
>         >         primary server. 
>         >         
>         >         Now that most of the drama has past, I'm back to
>         trying to get 
>         >         mod_skypopen 
>         >         to install. 
>         >         Before F17 I was running F14 so you may see why the
>         upgrade 
>         >         was required. 
>         >         'nough said, 
>         >         
>         >         Just like F14 I'm seeing an error in running the
>         install 
>         >         sequence 
>         >         
>         >
>         cd /usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss 
>         >         make clean; make; insmod ./skypopen.ko;
>         mknod /dev/dsp c 14 3 
>         >         
>         >         once again I see 
>         >         insmod: error inserting './skypopen.ko': -1 Device
>         or resource 
>         >         busy 
>         >         mknod: `/dev/dsp': File exists 
>         >         
>         >         I'm told previously that I don't want to see any snd
>         modules 
>         >         listed in the 
>         >         command 
>         >         
>         >         lsmod |grep snd 
>         >         
>         >         
>         >         but in my case I see several 
>         >         
>         >         lsmod | grep snd 
>         >         
>         >         snd_hda_codec_realtek   223962  1 
>         >         snd_hda_intel          33773  3 
>         >         snd_hda_codec         127706  2 
>         >         snd_hda_codec_realtek,snd_hda_intel 
>         >         snd_hwdep              13668  1 snd_hda_codec 
>         >         snd_pcm                97188  2
>         snd_hda_intel,snd_hda_codec 
>         >         snd_seq_midi           13324  0 
>         >         snd_rawmidi            30748  1 snd_seq_midi 
>         >         snd_seq_midi_event     14899  1 snd_seq_midi 
>         >         snd_seq                61896  2 
>         >         snd_seq_midi,snd_seq_midi_event 
>         >         snd_timer              29990  2 snd_pcm,snd_seq 
>         >         snd_seq_device         14540  3 
>         >         snd_seq_midi,snd_rawmidi,snd_seq 
>         >         snd                    78855  15 
>         >
>         snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device 
>         >         soundcore              15091  1 snd 
>         >         snd_page_alloc         18529  2
>         snd_hda_intel,snd_pcm 
>         >         
>         >         Questions: 
>         >         
>         >         do I need to rmmod each of these before I can
>         successfully 
>         >         install the 
>         >         mod_skypopen ? 
>         >         
>         >         Assuming this is true,  can I insmod the list to
>         return to the 
>         >         previous 
>         >         state or does the install of mod_skypopen limit what
>         I can do 
>         >         with the 
>         >         systems audio capabilities? 
>         >         I utilize other applications that require audio ie:
>         alsa and 
>         >         pulse modules 
>         >         to function. 
>         >         
>         >         Thanks 
>         >         
>         >         
>         >         
>         >         -- 
>         >         View this message in context: 
>         >
>         http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628.html
>         >         Sent from the freeswitch-users mailing list archive
>         at 
>         >         Nabble.com. 
>         >         
>         >
>         _________________________________________________________________________ 
>         >         Professional FreeSWITCH Consulting Services: 
>         
>         >         [hidden email] 
>         >         http://www.freeswitchsolutions.com
>         >         
>         >         FreeSWITCH-powered IP PBX: The CudaTel Communication
>         Server 
>         >         
>         >         
>         >         Official FreeSWITCH Sites 
>         >         http://www.freeswitch.org
>         >         http://wiki.freeswitch.org
>         >         http://www.cluecon.com
>         >         
>         >         Join Us At ClueCon - Aug 7-9, 2012 
>         >         
>         >         FreeSWITCH-users mailing list 
>         
>         >         [hidden email] 
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         >         http://www.freeswitch.org
>         > 
>         > 
>         > 
>         > -- 
>         > Sincerely, 
>         > 
>         > Giovanni Maruzzelli 
>         > Cell : <a href="tel:%2B39-347-2665618" value="+393472665618"
>         target="_blank">+39-347-2665618 
>         > 
>         > 
>         
>         >
>         _________________________________________________________________________ 
>         > Professional FreeSWITCH Consulting Services: 
>         
>         > [hidden email] 
>         > http://www.freeswitchsolutions.com
>         > 
>         >  
>         > 
>         > 
>         > Official FreeSWITCH Sites 
>         > http://www.freeswitch.org
>         > http://wiki.freeswitch.org
>         > http://www.cluecon.com
>         > 
>         > Join Us At ClueCon - Aug 7-9, 2012 
>         > 
>         > FreeSWITCH-users mailing list 
>         
>         > [hidden email] 
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>         > 
>         > 
>         > 
>         
>         >
>         ______________________________________________________________________ 
>         > If you reply to this email, your message will be added to
>         the 
>         > discussion below: 
>         >
>         http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628p7581631.html  
>         > To unsubscribe from Building Skype Issues, click here. 
>         > NAML 
>         
>         -- 
>         - 
>         - 
>         -    Best Regards, 
>         - 
>         -            Todd Bailey 
>         - 
>         - 
>         
>         
>         
>         ______________________________________________________________
>         View this message in context: Re: Building Skype Issues
>         
>         Sent from the freeswitch-users mailing list archive at
>         Nabble.com.
>         
>         
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         [hidden email]
>         http://www.freeswitchsolutions.com
>         
>         
>         
>         
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>         
>         Join Us At ClueCon - Aug 7-9, 2012
>         
>         FreeSWITCH-users mailing list
>         [hidden email]
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> 
> 
> _________________________________________________________________________ 
> Professional FreeSWITCH Consulting Services: 
> [hidden email] 
> http://www.freeswitchsolutions.com
> 
>  
> 
> 
> Official FreeSWITCH Sites 
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012 
> 
> FreeSWITCH-users mailing list 
> [hidden email] 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> ______________________________________________________________________
> If you reply to this email, your message will be added to the
> discussion below:
> http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628p7581636.html 
> To unsubscribe from Building Skype Issues, click here.
> NAML 

-- 
-
-
-    Best Regards,
-
-            Todd Bailey
-
-





--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628p7581638.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From ben122uk at gmail.com  Wed Aug  8 01:32:09 2012
From: ben122uk at gmail.com (Ben N)
Date: Tue, 7 Aug 2012 22:32:09 +0100
Subject: [Freeswitch-users] Different ptime causes call to fail
Message-ID: <CADaqzFPmxufq_WibeS_oQAmeFn4d6LUnsHhCdxsZ1V44CCdw1A@mail.gmail.com>

Hi All

Wondering if someone could help me out with a ptime issue I'm having?

Some background info, we have a SIP client that uses G729 at 20ms for Wi-Fi,
and G729 at 50ms for 3G.  The 3G call is the one I'm having trouble with.
 When a call is made from the SIP client to the PSTN via our carrier when a
ptime of 50ms is used, the call fails.  The carrier does support G729 with
variable ptimes.  I also think that the audio stream sent back from the
carrier is G729 50ms, I have confirmed this by analyzing the RTP stream in
Wireshark.

In SIP terms, Freeswitch receives a 183 with SDP from the carrier which
gets passed to the SIP client.  Freeswitch then gets a 200 OK from the
carrier, but Freeswitch does not send an ACK to the carrier for this.  When
20ms is used, the ACK is sent.  This could just be a by-product of a
different problem, but I've yet to pin-point what that is.

In my vars.xml, I have both global and outbound codecs set to "=G729 at 50i,G729",
I've also tried them the other way around but I don't think this makes a
difference.

I have late negotiation set to false on my internal profile, and disable
transcoding set to true so that whatever is dynamically used on  the A-leg,
gets used on the B-leg.

I'm not using absolute codec strings for G729 at 50i, as I would think this
make things fairly static, and then affects Wi-Fi calls that are trying to
use 20ms.  Out of curiosity, i set the absolute codec string to 50ms (
{absolute_codec_string=G729 at 50i} within bridge application) and the same
thing happened.

Oh and also I have my codec negotiation to "generous".

In terms of the call failing, what happens is that the PSTN device rings,
the SIP client goes to a ringing state, but when the PSTN device is picked
up the SIP client does not progress to the next step.  I have noticed that
the signalling by this point is still stuck on sending an ACK to the
carrier, so signalling probably isn't working anyway.

Finally, I have mod_com_g729 loaded with a couple of licenses on the
server.  The idea is to not use a license for this kind of call, as both
legs are supposed to be using G729 at 50i.  I believe this is happening too
when making a call, because I can't see and encoder/decoder being created
in the FS console.

Has anyone had G729 with ptime of 50ms working in passthru mode when using
mod_com_g729?  I

My Freeswitch version is about a year old now, but I would like to avoid
updating if possible.  I can provide console/SIP logs if required!

I really am stuck on this one, so thanks in advance for any help!  If this
turns out to be beyond the scope of this mailing list, then Freeswitch
consultation would not be out of the question.
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From mario_fs at mgtech.com  Wed Aug  8 01:36:04 2012
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 7 Aug 2012 14:36:04 -0700
Subject: [Freeswitch-users] Help with RECOVERY_ON_TIMER_EXPIRE?
Message-ID: <1E132B07-7B78-4CD1-B01E-63D079AFDFC6@mgtech.com>

I searched the web and mailing list but nothing matched. It occurs during ringing of Bria on an iPad, when I press the notification to answer the call this error occurs intermittently. Can anyone shed some light on this? Been on this problem for months now. Thanks,
Mario G 

2012-08-06 18:02:48.620334 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:110 at 11.135.1.10:56309) Running State Change CS_REPORTING
2012-08-06 18:02:48.620334 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:110 at 11.135.1.10:56309) State REPORTING
2012-08-06 18:02:48.620334 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:110 at 11.135.1.10:56309 Standard REPORTING, cause: RECOVERY_ON_TIMER_EXPIRE
2012-08-06 18:02:48.620334 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:110 at 11.135.1.10:56309) State REPORTING going to sleep
2012-08-06 18:02:48.620334 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:110 at 11.135.1.10:56309) State Change CS_REPORTING -> CS_DESTROY
2012-08-06 18:02:48.620334 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:110 at 11.135.1.10:56309 [BREAK]
2012-08-06 18:02:48.620334 [DEBUG] switch_core_session.c:1429 Session 76 (sofia/internal/sip:110 at 11.135.1.10:56309) Locked, Waiting

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From mitch.capper at gmail.com  Wed Aug  8 02:41:34 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 7 Aug 2012 15:41:34 -0700
Subject: [Freeswitch-users] Building Skype Issues
In-Reply-To: <1344373154.5586.10.camel@mythtv>
References: <1344369228639-7581628.post@n2.nabble.com>
	<CALXCt0ojh_AvsBLhDiaUHdq8PU-_fJtEXaoF9cfyC2JtASwh3Q@mail.gmail.com>
	<1344370768.5586.7.camel@mythtv>
	<CALXCt0o0b1YjjWe2+jFzitANkU=6fpKbEL4K9k=TgB1uiau-uw@mail.gmail.com>
	<1344373154.5586.10.camel@mythtv>
Message-ID: <CALn233pA29F8ebewJE++XXjVMhqQp=nju9WVnzsiL=cn_JdOSg@mail.gmail.com>

Skype does not allow 3rd party apps to do what giovanni has so
amazingly been able to get working quite smoothly with skypopen so
most of us don't consider the requirements too harsh.  You should not
be running a production PBX on a server running a desktop and that
also does audio playback.   The best solution is a standalone server a
virtual server may work but running audio through virtual servers is
not always the best experience.   In addition the host OS is still
running as a desktop which is not good to start.  Desktops and audio
playback are meant to be user preempted and ultra responsive where as
on a server you want to make sure everything has a fair time slice.

To answer the question feel free to try and hack around the
requirements for skyopen but they are stated (twice) by the author so
if you run into audio gliches or other problems beware that getting
support in less than likely in an unsupported environment.

~mitch

On Tue, Aug 7, 2012 at 1:59 PM, toddbailey <toddb at toddbailey.net> wrote:
> Sounds like skypopen needs to be expanded to play nice with others.
>
> Will creating a virtual machine and running a server command line only
> version of a "supported" os  resolve the snd module issues?
>
>
> On Tue, 2012-08-07 at 13:37 -0700, Giovanni Maruzzelli-2 [via
> freeswitch-users] wrote:
>
>>
>>
>> On Tue, Aug 7, 2012 at 10:19 PM, toddbailey <[hidden email]> wrote:
>>
>>         Mint 13 LST is a long term supported product based off Ubuntu
>>         12 so it
>>         should supported at least for the most part.  Any minor
>>         nuances I can
>>         deal with.
>>
>>         What exactly do you mean do not use a desktop distro ?
>>                 I need the gnome desktop for other activities.
>>
>>
>> a desktop will not work with skypopen, skypopen is designed to run on
>> a server distro.
>>
>> If you want to explore, try to understand what is starting the oss
>> (open sound system) alsa emulation on your system. Sorry, I don't know
>> which modules you'll have to unload. You probably need to modify the
>> settings of pulseaudio too.
>>
>> Ask in alsa and puleaudio forums how you can unload the OSS emulation.
>>
>> I repeat, you'll end up trashing a lot of time and be very frustrated.
>>
>> I will no more step up in this, use a server install - I told you
>> twice ;). (I'm the developer of skypopen)
>>
>> -giovanni
>>
>>
>>
>>
>>
>>
>>         yes, rmmod all the snd* (but, why are there?)
>>         Because other applications installed use sound modules.
>>
>>         This is a multipurpose server that provides many different
>>         functions, a
>>         pbx is just one of several functions.
>>
>>         Should I consider building a virtual machine and running free
>>         switch in
>>         it instead on the core os?
>>
>>
>>         On Tue, 2012-08-07 at 13:08 -0700, Giovanni Maruzzelli-2 [via
>>         freeswitch-users] wrote:
>>
>>         > you better follow the wikipage, that tells you to use centos
>>         6.x or
>>         > ubuntu, server installations, at 64 bit, or windows.
>>         >
>>         > If you choose to use a different OS, you need to know how to
>>         deal with
>>         > it.
>>         >
>>         > 1) do not use a desktop distro, it will not work
>>         > 2) yes, rmmod all the snd* (but, why are there? You want
>>         them not
>>         > loaded at all. Uloading is a hack)
>>         >
>>         > so, to avoid frustration, use a supported OS.
>>         >
>>         > In all other cases, you are going to use an awful amount of
>>         time and
>>         > get lot of frustrations, I can guarantee you ;).
>>         >
>>         > -giovanni
>>         >
>>         >
>>         >
>>         > On Tue, Aug 7, 2012 at 9:53 PM, toddbailey <[hidden email]>
>>         wrote:
>>
>>         >         Hi All,
>>         >
>>         >         After a dreadful several weeks of trying to get my
>>         Fedora 17
>>         >         upgrade from
>>         >         F14 to cooperate, I'm now running Mint 13 LST x64 as
>>         my
>>         >         primary server.
>>         >
>>         >         Now that most of the drama has past, I'm back to
>>         trying to get
>>         >         mod_skypopen
>>         >         to install.
>>         >         Before F17 I was running F14 so you may see why the
>>         upgrade
>>         >         was required.
>>         >         'nough said,
>>         >
>>         >         Just like F14 I'm seeing an error in running the
>>         install
>>         >         sequence
>>         >
>>         >
>>         cd /usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss
>>         >         make clean; make; insmod ./skypopen.ko;
>>         mknod /dev/dsp c 14 3
>>         >
>>         >         once again I see
>>         >         insmod: error inserting './skypopen.ko': -1 Device
>>         or resource
>>         >         busy
>>         >         mknod: `/dev/dsp': File exists
>>         >
>>         >         I'm told previously that I don't want to see any snd
>>         modules
>>         >         listed in the
>>         >         command
>>         >
>>         >         lsmod |grep snd
>>         >
>>         >
>>         >         but in my case I see several
>>         >
>>         >         lsmod | grep snd
>>         >
>>         >         snd_hda_codec_realtek   223962  1
>>         >         snd_hda_intel          33773  3
>>         >         snd_hda_codec         127706  2
>>         >         snd_hda_codec_realtek,snd_hda_intel
>>         >         snd_hwdep              13668  1 snd_hda_codec
>>         >         snd_pcm                97188  2
>>         snd_hda_intel,snd_hda_codec
>>         >         snd_seq_midi           13324  0
>>         >         snd_rawmidi            30748  1 snd_seq_midi
>>         >         snd_seq_midi_event     14899  1 snd_seq_midi
>>         >         snd_seq                61896  2
>>         >         snd_seq_midi,snd_seq_midi_event
>>         >         snd_timer              29990  2 snd_pcm,snd_seq
>>         >         snd_seq_device         14540  3
>>         >         snd_seq_midi,snd_rawmidi,snd_seq
>>         >         snd                    78855  15
>>         >
>>
>> snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device
>>         >         soundcore              15091  1 snd
>>         >         snd_page_alloc         18529  2
>>         snd_hda_intel,snd_pcm
>>         >
>>         >         Questions:
>>         >
>>         >         do I need to rmmod each of these before I can
>>         successfully
>>         >         install the
>>         >         mod_skypopen ?
>>         >
>>         >         Assuming this is true,  can I insmod the list to
>>         return to the
>>         >         previous
>>         >         state or does the install of mod_skypopen limit what
>>         I can do
>>         >         with the
>>         >         systems audio capabilities?
>>         >         I utilize other applications that require audio ie:
>>         alsa and
>>         >         pulse modules
>>         >         to function.
>>         >
>>         >         Thanks
>>         >
>>         >
>>         >
>>         >         --
>>         >         View this message in context:
>>         >
>>
>> http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628.html
>>         >         Sent from the freeswitch-users mailing list archive
>>         at
>>         >         Nabble.com.
>>         >
>>         >
>>
>> _________________________________________________________________________
>>         >         Professional FreeSWITCH Consulting Services:
>>
>>         >         [hidden email]
>>         >         http://www.freeswitchsolutions.com
>>         >
>>         >         FreeSWITCH-powered IP PBX: The CudaTel Communication
>>         Server
>>         >         
>>         >
>>         >         Official FreeSWITCH Sites
>>         >         http://www.freeswitch.org
>>         >         http://wiki.freeswitch.org
>>         >         http://www.cluecon.com
>>         >
>>         >         Join Us At ClueCon - Aug 7-9, 2012
>>         >
>>         >         FreeSWITCH-users mailing list
>>
>>         >         [hidden email]
>>         >
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         >
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         >         http://www.freeswitch.org
>>         >
>>         >
>>         >
>>         > --
>>         > Sincerely,
>>         >
>>         > Giovanni Maruzzelli
>>         > Cell : <a href="tel:%2B39-347-2665618" value="+393472665618"
>>         target="_blank">+39-347-2665618
>>         >
>>         >
>>
>>         >
>>
>> _________________________________________________________________________
>>         > Professional FreeSWITCH Consulting Services:
>>
>>         > [hidden email]
>>         > http://www.freeswitchsolutions.com
>>         >
>>         > 
>>         > 
>>         >
>>         > Official FreeSWITCH Sites
>>         > http://www.freeswitch.org
>>         > http://wiki.freeswitch.org
>>         > http://www.cluecon.com
>>         >
>>         > Join Us At ClueCon - Aug 7-9, 2012
>>         >
>>         > FreeSWITCH-users mailing list
>>
>>         > [hidden email]
>>         >
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         >
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         > http://www.freeswitch.org
>>         >
>>         >
>>         >
>>
>>         >
>>
>> ______________________________________________________________________
>>         > If you reply to this email, your message will be added to
>>         the
>>         > discussion below:
>>         >
>>
>> http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628p7581631.html
>>         > To unsubscribe from Building Skype Issues, click here.
>>         > NAML
>>
>>         --
>>         -
>>         -
>>         -    Best Regards,
>>         -
>>         -            Todd Bailey
>>         -
>>         -
>>
>>
>>
>>         ______________________________________________________________
>>         View this message in context: Re: Building Skype Issues
>>
>>         Sent from the freeswitch-users mailing list archive at
>>         Nabble.com.
>>
>>
>>
>> _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         [hidden email]
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.com
>>
>>         Join Us At ClueCon - Aug 7-9, 2012
>>
>>         FreeSWITCH-users mailing list
>>         [hidden email]
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> [hidden email]
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> [hidden email]
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> ______________________________________________________________________
>> If you reply to this email, your message will be added to the
>> discussion below:
>>
>> http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628p7581636.html
>> To unsubscribe from Building Skype Issues, click here.
>> NAML
>
> --
> -
> -
> -    Best Regards,
> -
> -            Todd Bailey
> -
> -
>
>
> ________________________________
> View this message in context: Re: Building Skype Issues
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mitch.capper at gmail.com  Wed Aug  8 02:42:27 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 7 Aug 2012 15:42:27 -0700
Subject: [Freeswitch-users] Building Skype Issues
In-Reply-To: <CALn233pA29F8ebewJE++XXjVMhqQp=nju9WVnzsiL=cn_JdOSg@mail.gmail.com>
References: <1344369228639-7581628.post@n2.nabble.com>
	<CALXCt0ojh_AvsBLhDiaUHdq8PU-_fJtEXaoF9cfyC2JtASwh3Q@mail.gmail.com>
	<1344370768.5586.7.camel@mythtv>
	<CALXCt0o0b1YjjWe2+jFzitANkU=6fpKbEL4K9k=TgB1uiau-uw@mail.gmail.com>
	<1344373154.5586.10.camel@mythtv>
	<CALn233pA29F8ebewJE++XXjVMhqQp=nju9WVnzsiL=cn_JdOSg@mail.gmail.com>
Message-ID: <CALn233ryW7o4dOnavOUMcqzTCaS-o1TLBGjkRiym6uO9i0pvMw@mail.gmail.com>

Sorry also keep i mind that FS welcomes development so if you do make
it work in other environments or want to put a bounty towards your
requirements its always appreciated.

~Mitch

On Tue, Aug 7, 2012 at 3:41 PM, Mitch Capper <mitch.capper at gmail.com> wrote:
> Skype does not allow 3rd party apps to do what giovanni has so
> amazingly been able to get working quite smoothly with skypopen so
> most of us don't consider the requirements too harsh.  You should not
> be running a production PBX on a server running a desktop and that
> also does audio playback.   The best solution is a standalone server a
> virtual server may work but running audio through virtual servers is
> not always the best experience.   In addition the host OS is still
> running as a desktop which is not good to start.  Desktops and audio
> playback are meant to be user preempted and ultra responsive where as
> on a server you want to make sure everything has a fair time slice.
>
> To answer the question feel free to try and hack around the
> requirements for skyopen but they are stated (twice) by the author so
> if you run into audio gliches or other problems beware that getting
> support in less than likely in an unsupported environment.
>
> ~mitch
>
> On Tue, Aug 7, 2012 at 1:59 PM, toddbailey <toddb at toddbailey.net> wrote:
>> Sounds like skypopen needs to be expanded to play nice with others.
>>
>> Will creating a virtual machine and running a server command line only
>> version of a "supported" os  resolve the snd module issues?
>>
>>
>> On Tue, 2012-08-07 at 13:37 -0700, Giovanni Maruzzelli-2 [via
>> freeswitch-users] wrote:
>>
>>>
>>>
>>> On Tue, Aug 7, 2012 at 10:19 PM, toddbailey <[hidden email]> wrote:
>>>
>>>         Mint 13 LST is a long term supported product based off Ubuntu
>>>         12 so it
>>>         should supported at least for the most part.  Any minor
>>>         nuances I can
>>>         deal with.
>>>
>>>         What exactly do you mean do not use a desktop distro ?
>>>                 I need the gnome desktop for other activities.
>>>
>>>
>>> a desktop will not work with skypopen, skypopen is designed to run on
>>> a server distro.
>>>
>>> If you want to explore, try to understand what is starting the oss
>>> (open sound system) alsa emulation on your system. Sorry, I don't know
>>> which modules you'll have to unload. You probably need to modify the
>>> settings of pulseaudio too.
>>>
>>> Ask in alsa and puleaudio forums how you can unload the OSS emulation.
>>>
>>> I repeat, you'll end up trashing a lot of time and be very frustrated.
>>>
>>> I will no more step up in this, use a server install - I told you
>>> twice ;). (I'm the developer of skypopen)
>>>
>>> -giovanni
>>>
>>>
>>>
>>>
>>>
>>>
>>>         yes, rmmod all the snd* (but, why are there?)
>>>         Because other applications installed use sound modules.
>>>
>>>         This is a multipurpose server that provides many different
>>>         functions, a
>>>         pbx is just one of several functions.
>>>
>>>         Should I consider building a virtual machine and running free
>>>         switch in
>>>         it instead on the core os?
>>>
>>>
>>>         On Tue, 2012-08-07 at 13:08 -0700, Giovanni Maruzzelli-2 [via
>>>         freeswitch-users] wrote:
>>>
>>>         > you better follow the wikipage, that tells you to use centos
>>>         6.x or
>>>         > ubuntu, server installations, at 64 bit, or windows.
>>>         >
>>>         > If you choose to use a different OS, you need to know how to
>>>         deal with
>>>         > it.
>>>         >
>>>         > 1) do not use a desktop distro, it will not work
>>>         > 2) yes, rmmod all the snd* (but, why are there? You want
>>>         them not
>>>         > loaded at all. Uloading is a hack)
>>>         >
>>>         > so, to avoid frustration, use a supported OS.
>>>         >
>>>         > In all other cases, you are going to use an awful amount of
>>>         time and
>>>         > get lot of frustrations, I can guarantee you ;).
>>>         >
>>>         > -giovanni
>>>         >
>>>         >
>>>         >
>>>         > On Tue, Aug 7, 2012 at 9:53 PM, toddbailey <[hidden email]>
>>>         wrote:
>>>
>>>         >         Hi All,
>>>         >
>>>         >         After a dreadful several weeks of trying to get my
>>>         Fedora 17
>>>         >         upgrade from
>>>         >         F14 to cooperate, I'm now running Mint 13 LST x64 as
>>>         my
>>>         >         primary server.
>>>         >
>>>         >         Now that most of the drama has past, I'm back to
>>>         trying to get
>>>         >         mod_skypopen
>>>         >         to install.
>>>         >         Before F17 I was running F14 so you may see why the
>>>         upgrade
>>>         >         was required.
>>>         >         'nough said,
>>>         >
>>>         >         Just like F14 I'm seeing an error in running the
>>>         install
>>>         >         sequence
>>>         >
>>>         >
>>>         cd /usr/local/src/freeswitch/src/mod/endpoints/mod_skypopen/oss
>>>         >         make clean; make; insmod ./skypopen.ko;
>>>         mknod /dev/dsp c 14 3
>>>         >
>>>         >         once again I see
>>>         >         insmod: error inserting './skypopen.ko': -1 Device
>>>         or resource
>>>         >         busy
>>>         >         mknod: `/dev/dsp': File exists
>>>         >
>>>         >         I'm told previously that I don't want to see any snd
>>>         modules
>>>         >         listed in the
>>>         >         command
>>>         >
>>>         >         lsmod |grep snd
>>>         >
>>>         >
>>>         >         but in my case I see several
>>>         >
>>>         >         lsmod | grep snd
>>>         >
>>>         >         snd_hda_codec_realtek   223962  1
>>>         >         snd_hda_intel          33773  3
>>>         >         snd_hda_codec         127706  2
>>>         >         snd_hda_codec_realtek,snd_hda_intel
>>>         >         snd_hwdep              13668  1 snd_hda_codec
>>>         >         snd_pcm                97188  2
>>>         snd_hda_intel,snd_hda_codec
>>>         >         snd_seq_midi           13324  0
>>>         >         snd_rawmidi            30748  1 snd_seq_midi
>>>         >         snd_seq_midi_event     14899  1 snd_seq_midi
>>>         >         snd_seq                61896  2
>>>         >         snd_seq_midi,snd_seq_midi_event
>>>         >         snd_timer              29990  2 snd_pcm,snd_seq
>>>         >         snd_seq_device         14540  3
>>>         >         snd_seq_midi,snd_rawmidi,snd_seq
>>>         >         snd                    78855  15
>>>         >
>>>
>>> snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device
>>>         >         soundcore              15091  1 snd
>>>         >         snd_page_alloc         18529  2
>>>         snd_hda_intel,snd_pcm
>>>         >
>>>         >         Questions:
>>>         >
>>>         >         do I need to rmmod each of these before I can
>>>         successfully
>>>         >         install the
>>>         >         mod_skypopen ?
>>>         >
>>>         >         Assuming this is true,  can I insmod the list to
>>>         return to the
>>>         >         previous
>>>         >         state or does the install of mod_skypopen limit what
>>>         I can do
>>>         >         with the
>>>         >         systems audio capabilities?
>>>         >         I utilize other applications that require audio ie:
>>>         alsa and
>>>         >         pulse modules
>>>         >         to function.
>>>         >
>>>         >         Thanks
>>>         >
>>>         >
>>>         >
>>>         >         --
>>>         >         View this message in context:
>>>         >
>>>
>>> http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628.html
>>>         >         Sent from the freeswitch-users mailing list archive
>>>         at
>>>         >         Nabble.com.
>>>         >
>>>         >
>>>
>>> _________________________________________________________________________
>>>         >         Professional FreeSWITCH Consulting Services:
>>>
>>>         >         [hidden email]
>>>         >         http://www.freeswitchsolutions.com
>>>         >
>>>         >         FreeSWITCH-powered IP PBX: The CudaTel Communication
>>>         Server
>>>         >         
>>>         >
>>>         >         Official FreeSWITCH Sites
>>>         >         http://www.freeswitch.org
>>>         >         http://wiki.freeswitch.org
>>>         >         http://www.cluecon.com
>>>         >
>>>         >         Join Us At ClueCon - Aug 7-9, 2012
>>>         >
>>>         >         FreeSWITCH-users mailing list
>>>
>>>         >         [hidden email]
>>>         >
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         >
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         >         http://www.freeswitch.org
>>>         >
>>>         >
>>>         >
>>>         > --
>>>         > Sincerely,
>>>         >
>>>         > Giovanni Maruzzelli
>>>         > Cell : <a href="tel:%2B39-347-2665618" value="+393472665618"
>>>         target="_blank">+39-347-2665618
>>>         >
>>>         >
>>>
>>>         >
>>>
>>> _________________________________________________________________________
>>>         > Professional FreeSWITCH Consulting Services:
>>>
>>>         > [hidden email]
>>>         > http://www.freeswitchsolutions.com
>>>         >
>>>         > 
>>>         > 
>>>         >
>>>         > Official FreeSWITCH Sites
>>>         > http://www.freeswitch.org
>>>         > http://wiki.freeswitch.org
>>>         > http://www.cluecon.com
>>>         >
>>>         > Join Us At ClueCon - Aug 7-9, 2012
>>>         >
>>>         > FreeSWITCH-users mailing list
>>>
>>>         > [hidden email]
>>>         >
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         >
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         > http://www.freeswitch.org
>>>         >
>>>         >
>>>         >
>>>
>>>         >
>>>
>>> ______________________________________________________________________
>>>         > If you reply to this email, your message will be added to
>>>         the
>>>         > discussion below:
>>>         >
>>>
>>> http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628p7581631.html
>>>         > To unsubscribe from Building Skype Issues, click here.
>>>         > NAML
>>>
>>>         --
>>>         -
>>>         -
>>>         -    Best Regards,
>>>         -
>>>         -            Todd Bailey
>>>         -
>>>         -
>>>
>>>
>>>
>>>         ______________________________________________________________
>>>         View this message in context: Re: Building Skype Issues
>>>
>>>         Sent from the freeswitch-users mailing list archive at
>>>         Nabble.com.
>>>
>>>
>>>
>>> _________________________________________________________________________
>>>         Professional FreeSWITCH Consulting Services:
>>>         [hidden email]
>>>         http://www.freeswitchsolutions.com
>>>
>>>         
>>>         
>>>
>>>         Official FreeSWITCH Sites
>>>         http://www.freeswitch.org
>>>         http://wiki.freeswitch.org
>>>         http://www.cluecon.com
>>>
>>>         Join Us At ClueCon - Aug 7-9, 2012
>>>
>>>         FreeSWITCH-users mailing list
>>>         [hidden email]
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> [hidden email]
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> [hidden email]
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> ______________________________________________________________________
>>> If you reply to this email, your message will be added to the
>>> discussion below:
>>>
>>> http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628p7581636.html
>>> To unsubscribe from Building Skype Issues, click here.
>>> NAML
>>
>> --
>> -
>> -
>> -    Best Regards,
>> -
>> -            Todd Bailey
>> -
>> -
>>
>>
>> ________________________________
>> View this message in context: Re: Building Skype Issues
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>


From mitch.capper at gmail.com  Wed Aug  8 02:44:30 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 7 Aug 2012 15:44:30 -0700
Subject: [Freeswitch-users] upadte FS (make current)
In-Reply-To: <1FFF97C269757C458224B7C895F35F1513FFDD@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1513F759@cantor.std.visionutv.se>
	<5020EC2A.2080302@softnet.si> <web-86139963@softnet.si>
	<1FFF97C269757C458224B7C895F35F1513FFDD@cantor.std.visionutv.se>
Message-ID: <CALn233rh9GL7_=o2nyhi4RSuGi6S=x8JfUnWxJdZ5J4Y6HSreA@mail.gmail.com>

Linux memory can be a bit tricky to always track well the application
ATOP allows you sort by memory usage and is pretty good.   Keep in
mind however you may see the free memory going down on the system but
really its just used for file caching or what have you so its not
really lost or not available.  Linux doesn't always do an easy job of
showing memory usage.

~Mitch

On Tue, Aug 7, 2012 at 12:51 PM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Why do you even think FS leaks memory? How much memory does it use?
>
> Anyway, to look for memory leaks, check out this:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Collection_Information_With_Valgrind_.28Linux.2FUnix.29
>
> /Peter
>
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Miha [miha at softnet.si]
> Skickat: den 7 augusti 2012 21:22
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] upadte FS (make current)
>
> HI,
>
> just one more information. What is the best way to see what
> leaks memory and to be sure that FS does not leak it?
>
> If I look at the top command I can see that the free memory
> is slowly dropping, but from ps aux I can see that FS
> consums 1.3% of memory (the same).
>
> Thanks!
>
> Miha
>
> Thanks!
>  MIha
>
> On Tue, 07 Aug 2012 12:21:30 +0200
>  Miha <miha at softnet.si> wrote:
>> Hi Peter,
>>
>> fsctl reclaim_mem this will not harm any call and etc?
>>
>> So if freeswitch does not leak memory what should I see
>> in top (memory will not drop?)?
>>
>> Thanks!
>> Miha
>>
>>
>>
>> On 8/7/2012 11:48 AM, Peter Olsson wrote:
>> > It's always hard to say what is a memory leak and
>> what's not. FS uses memory pools internally, which
>> sometime might make it look like it leaks memory, but in
>> the end FS will reuse that memory. You can also run
>> "fsctl reclaim_mem" in the console or fs_cli, and see if
>> that makes the memory usage drop.
>> >
>> > /Peter
>> >
>> > -----Ursprungligt meddelande-----
>> > Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
>> F?r Miha
>> > Skickat: den 7 augusti 2012 11:14
>> > Till: FreeSWITCH Users Help
>> > ?mne: Re: [Freeswitch-users] upadte FS (make current)
>> >
>> > On 8/7/2012 10:54 AM, Peter Olsson wrote:
>> >> Usually it's no problem. However, I always recommend
>> to have a test
>> >> server to run basic tests before going into
>> production. There have
>> >> been a couple of memory leaks fixed, but I think that
>> was before your
>> >> current version - but I don't remember all the commits
>> from my head..
>> >> :)
>> >>
>> >> I would also suggest to read through the git commit
>> log, to get an idea of what has been done.
>> >>
>> >> /Peter
>> >>
>> >>
>> >> -----Ursprungligt meddelande-----
>> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
>> F?r Miha
>> >> Skickat: den 7 augusti 2012 10:37
>> >> Till: FreeSWITCH Users Help
>> >> ?mne: [Freeswitch-users] upadte FS (make current)
>> >>
>> >> Hi,
>> >>
>> >> I am running on production server FS (runs ok). I
>> would like to upgrade it to latest git.
>> >> Can I expect any problems? I not using any special
>> module (just mod_rad_auth, for cdr mod_xml_cdr).
>> >>
>> >> Is it better to first test everything on test server
>> and than make upgrade?
>> >>
>> >> My FS is now up for 13 days. I am looking at memory
>> for a few days and noticing that free memory is shrinking
>> (slowly). Was there any problem with memory leak? :
>> >>
>> >> top - 09:46:46 up 13 days, 12:32,  3 users,  load
>> average: 0.00, 0.04, 0.05
>> >> Tasks: 159 total,   1 running, 158 sleeping,   0
>> stopped,   0 zombie
>> >> Cpu(s):  0.6%us,  0.4%sy,  0.0%ni, 99.0%id,  0.0%wa,
>>  0.0%hi, 0.0%si, 0.0%st
>> >> Mem:   4046636k total,  3356340k used,   690296k free,
>>   357296k buffers
>> >> Swap:  6094840k total,        0k used,  6094840k free,
>>  2629280k cached
>> >>
>> >> My version: FreeSWITCH Version
>> >> 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5
>> >> (1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul
>> 2012 15:18:27 Z)
>> >>
>> >> Thanks!
>> >>
>> >> Miha
>> >>
>> >>
>>
> ______________________________________________________________________
>> >> ___ Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> Join Us At ClueCon - Aug 7-9, 2012
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>
>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> >> rs
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >>
>> >>
>>
> ______________________________________________________________________
>> >> ___ Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> Join Us At ClueCon - Aug 7-9, 2012
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>
>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> >> rs
>> >> http://www.freeswitch.org
>> >>
>> > @Peter thanks for your quick answer:)
>> >
>> > I readed jira postes and noticed that this was befor my
>> git version (as you said).
>> > If there was leak problem (I do not know if it is, just
>> noticing that free memory is droping), would FS be stable
>> for 13 days or would this bug be present sooner (memory
>> leak)?
>> >
>> > regards,
>> > Miha
>> >
>> >
>> >
>>
> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > Join Us At ClueCon - Aug 7-9, 2012
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> >
>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> >
>> >
>> >
>>
> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > Join Us At ClueCon - Aug 7-9, 2012
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> >
>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:5021693632761271914539!
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From toddb at toddbailey.net  Wed Aug  8 03:45:44 2012
From: toddb at toddbailey.net (toddbailey)
Date: Tue, 7 Aug 2012 16:45:44 -0700 (PDT)
Subject: [Freeswitch-users] Building Skype Issues
In-Reply-To: <CALn233ryW7o4dOnavOUMcqzTCaS-o1TLBGjkRiym6uO9i0pvMw@mail.gmail.com>
References: <1344369228639-7581628.post@n2.nabble.com>
	<CALXCt0ojh_AvsBLhDiaUHdq8PU-_fJtEXaoF9cfyC2JtASwh3Q@mail.gmail.com>
	<1344370768.5586.7.camel@mythtv>
	<CALXCt0o0b1YjjWe2+jFzitANkU=6fpKbEL4K9k=TgB1uiau-uw@mail.gmail.com>
	<1344373154.5586.10.camel@mythtv>
	<CALn233pA29F8ebewJE++XXjVMhqQp=nju9WVnzsiL=cn_JdOSg@mail.gmail.com>
	<CALn233ryW7o4dOnavOUMcqzTCaS-o1TLBGjkRiym6uO9i0pvMw@mail.gmail.com>
Message-ID: <1344383144276-7581644.post@n2.nabble.com>

No disrespect intended in my previous posts,  I applaud the dev team for what
they done,
I must have missed the paragraph that explicitly states you have to use a
dedicated server install in order to use mod_skypopen.    
I hope that I can use gnome for the initiial config and build process then
switch to run level 3 for normal use..
Not knowing that a server install was a requirement, I assumed I could
basically use any distro and any desktop environment.  FS works well with
out skype mods in a desktop env,  Maybe at some point in the future
mod_skypopen will see advances to allow it to co-habitate and operate and
any environment.

Anyway, 
After a frustrating several weeks of F17 I chose Mint 13 as a long term
stable (or soon to be) environment.
I had no idea that there would be audio issues to deal with. Due power
costs, I'm not overly excited about bringing another box online 24x7,  after
all I'm only running 1/2  dozen phone extensions and the primary server has
more than enough resources to support the additional load.  I just want to
be able to use multiple skype accounts to support multiple user's
extensions.

So I'm electing to build a VM that will only run FS. As suggested I'm
installing Centos 6.x (6.3 to be precise)
I'll do a full install sans any media apps and just enough to set network
parameters, a tftp server, and any dev tools to build FS.  Once every thing
is in place hopefully I can set Vbox to automatically boot and launch the
centos vm  with out any manual intervention.   well see how that works
out... eh?




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Building-Skype-Issues-tp7581628p7581644.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From miha at softnet.si  Wed Aug  8 10:12:13 2012
From: miha at softnet.si (Miha)
Date: Wed, 08 Aug 2012 08:12:13 +0200
Subject: [Freeswitch-users] upadte FS (make current)
In-Reply-To: <1FFF97C269757C458224B7C895F35F1513FFDD@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1513F759@cantor.std.visionutv.se>
	<5020EC2A.2080302@softnet.si>, <web-86139963@softnet.si>
	<1FFF97C269757C458224B7C895F35F1513FFDD@cantor.std.visionutv.se>
Message-ID: <5022033D.4090209@softnet.si>

On 8/7/2012 9:51 PM, Peter Olsson wrote:
> Why do you even think FS leaks memory? How much memory does it use?
>
> Anyway, to look for memory leaks, check out this:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Collection_Information_With_Valgrind_.28Linux.2FUnix.29
>
> /Peter
>
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Miha [miha at softnet.si]
> Skickat: den 7 augusti 2012 21:22
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] upadte FS (make current)
>
> HI,
>
> just one more information. What is the best way to see what
> leaks memory and to be sure that FS does not leak it?
>
> If I look at the top command I can see that the free memory
> is slowly dropping, but from ps aux I can see that FS
> consums 1.3% of memory (the same).
>
> Thanks!
>
> Miha
>
> Thanks!
>   MIha
>
> On Tue, 07 Aug 2012 12:21:30 +0200
>   Miha <miha at softnet.si> wrote:
>> Hi Peter,
>>
>> fsctl reclaim_mem this will not harm any call and etc?
>>
>> So if freeswitch does not leak memory what should I see
>> in top (memory will not drop?)?
>>
>> Thanks!
>> Miha
>>
>>
>>
>> On 8/7/2012 11:48 AM, Peter Olsson wrote:
>>> It's always hard to say what is a memory leak and
>> what's not. FS uses memory pools internally, which
>> sometime might make it look like it leaks memory, but in
>> the end FS will reuse that memory. You can also run
>> "fsctl reclaim_mem" in the console or fs_cli, and see if
>> that makes the memory usage drop.
>>> /Peter
>>>
>>> -----Ursprungligt meddelande-----
>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
>> F?r Miha
>>> Skickat: den 7 augusti 2012 11:14
>>> Till: FreeSWITCH Users Help
>>> ?mne: Re: [Freeswitch-users] upadte FS (make current)
>>>
>>> On 8/7/2012 10:54 AM, Peter Olsson wrote:
>>>> Usually it's no problem. However, I always recommend
>> to have a test
>>>> server to run basic tests before going into
>> production. There have
>>>> been a couple of memory leaks fixed, but I think that
>> was before your
>>>> current version - but I don't remember all the commits
>> from my head..
>>>> :)
>>>>
>>>> I would also suggest to read through the git commit
>> log, to get an idea of what has been done.
>>>> /Peter
>>>>
>>>>
>>>> -----Ursprungligt meddelande-----
>>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
>> F?r Miha
>>>> Skickat: den 7 augusti 2012 10:37
>>>> Till: FreeSWITCH Users Help
>>>> ?mne: [Freeswitch-users] upadte FS (make current)
>>>>
>>>> Hi,
>>>>
>>>> I am running on production server FS (runs ok). I
>> would like to upgrade it to latest git.
>>>> Can I expect any problems? I not using any special
>> module (just mod_rad_auth, for cdr mod_xml_cdr).
>>>> Is it better to first test everything on test server
>> and than make upgrade?
>>>> My FS is now up for 13 days. I am looking at memory
>> for a few days and noticing that free memory is shrinking
>> (slowly). Was there any problem with memory leak? :
>>>> top - 09:46:46 up 13 days, 12:32,  3 users,  load
>> average: 0.00, 0.04, 0.05
>>>> Tasks: 159 total,   1 running, 158 sleeping,   0
>> stopped,   0 zombie
>>>> Cpu(s):  0.6%us,  0.4%sy,  0.0%ni, 99.0%id,  0.0%wa,
>>   0.0%hi, 0.0%si, 0.0%st
>>>> Mem:   4046636k total,  3356340k used,   690296k free,
>>    357296k buffers
>>>> Swap:  6094840k total,        0k used,  6094840k free,
>>   2629280k cached
>>>> My version: FreeSWITCH Version
>>>> 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5
>>>> (1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul
>> 2012 15:18:27 Z)
>>>> Thanks!
>>>>
>>>> Miha
>>>>
>>>>
> ______________________________________________________________________
>>>> ___ Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>>> rs
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>
> ______________________________________________________________________
>>>> ___ Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>>> rs
>>>> http://www.freeswitch.org
>>>>
>>> @Peter thanks for your quick answer:)
>>>
>>> I readed jira postes and noticed that this was befor my
>> git version (as you said).
>>> If there was leak problem (I do not know if it is, just
>> noticing that free memory is droping), would FS be stable
>> for 13 days or would this bug be present sooner (memory
>> leak)?
>>> regards,
>>> Miha
>>>
>>>
>>>
> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>
> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:5021693632761271914539!
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
HI Peter,

I am not thinking that FS memory leaks, I just wont to be shure that my 
does not leaks:)

Problem is that I am notiching that in top command I can see that memory 
is slowly shrinking.

For exp in top command yesterday free memory was something about 635348k 
free

Today:

Mem:   4046636k total,  3511288k used,   535348k free,   367408k buffers
Swap:  6094840k total,        0k used,  6094840k free,  2765076k cached

So every day is a bit litle less.

ps aux:

root      3596  3.9  1.4 457624 58228 ?        S<l  Jul24 827:56 
/usr/local/freeswitch/bin/freeswitch -nc

As this machin is in production and I am going to vacation this week I 
do not wont to be any problems.

Regards,
Miha


From miha at softnet.si  Wed Aug  8 10:24:23 2012
From: miha at softnet.si (Miha)
Date: Wed, 08 Aug 2012 08:24:23 +0200
Subject: [Freeswitch-users] upadte FS (make current)
In-Reply-To: <CALn233rh9GL7_=o2nyhi4RSuGi6S=x8JfUnWxJdZ5J4Y6HSreA@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F1513F759@cantor.std.visionutv.se>
	<5020EC2A.2080302@softnet.si> <web-86139963@softnet.si>
	<1FFF97C269757C458224B7C895F35F1513FFDD@cantor.std.visionutv.se>
	<CALn233rh9GL7_=o2nyhi4RSuGi6S=x8JfUnWxJdZ5J4Y6HSreA@mail.gmail.com>
Message-ID: <50220617.9050306@softnet.si>

On 8/8/2012 12:44 AM, Mitch Capper wrote:
> Linux memory can be a bit tricky to always track well the application
> ATOP allows you sort by memory usage and is pretty good.   Keep in
> mind however you may see the free memory going down on the system but
> really its just used for file caching or what have you so its not
> really lost or not available.  Linux doesn't always do an easy job of
> showing memory usage.
>
> ~Mitch
>
> On Tue, Aug 7, 2012 at 12:51 PM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
>> Why do you even think FS leaks memory? How much memory does it use?
>>
>> Anyway, to look for memory leaks, check out this:
>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Collection_Information_With_Valgrind_.28Linux.2FUnix.29
>>
>> /Peter
>>
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Miha [miha at softnet.si]
>> Skickat: den 7 augusti 2012 21:22
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] upadte FS (make current)
>>
>> HI,
>>
>> just one more information. What is the best way to see what
>> leaks memory and to be sure that FS does not leak it?
>>
>> If I look at the top command I can see that the free memory
>> is slowly dropping, but from ps aux I can see that FS
>> consums 1.3% of memory (the same).
>>
>> Thanks!
>>
>> Miha
>>
>> Thanks!
>>   MIha
>>
>> On Tue, 07 Aug 2012 12:21:30 +0200
>>   Miha <miha at softnet.si> wrote:
>>> Hi Peter,
>>>
>>> fsctl reclaim_mem this will not harm any call and etc?
>>>
>>> So if freeswitch does not leak memory what should I see
>>> in top (memory will not drop?)?
>>>
>>> Thanks!
>>> Miha
>>>
>>>
>>>
>>> On 8/7/2012 11:48 AM, Peter Olsson wrote:
>>>> It's always hard to say what is a memory leak and
>>> what's not. FS uses memory pools internally, which
>>> sometime might make it look like it leaks memory, but in
>>> the end FS will reuse that memory. You can also run
>>> "fsctl reclaim_mem" in the console or fs_cli, and see if
>>> that makes the memory usage drop.
>>>> /Peter
>>>>
>>>> -----Ursprungligt meddelande-----
>>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
>>> F?r Miha
>>>> Skickat: den 7 augusti 2012 11:14
>>>> Till: FreeSWITCH Users Help
>>>> ?mne: Re: [Freeswitch-users] upadte FS (make current)
>>>>
>>>> On 8/7/2012 10:54 AM, Peter Olsson wrote:
>>>>> Usually it's no problem. However, I always recommend
>>> to have a test
>>>>> server to run basic tests before going into
>>> production. There have
>>>>> been a couple of memory leaks fixed, but I think that
>>> was before your
>>>>> current version - but I don't remember all the commits
>>> from my head..
>>>>> :)
>>>>>
>>>>> I would also suggest to read through the git commit
>>> log, to get an idea of what has been done.
>>>>> /Peter
>>>>>
>>>>>
>>>>> -----Ursprungligt meddelande-----
>>>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
>>> F?r Miha
>>>>> Skickat: den 7 augusti 2012 10:37
>>>>> Till: FreeSWITCH Users Help
>>>>> ?mne: [Freeswitch-users] upadte FS (make current)
>>>>>
>>>>> Hi,
>>>>>
>>>>> I am running on production server FS (runs ok). I
>>> would like to upgrade it to latest git.
>>>>> Can I expect any problems? I not using any special
>>> module (just mod_rad_auth, for cdr mod_xml_cdr).
>>>>> Is it better to first test everything on test server
>>> and than make upgrade?
>>>>> My FS is now up for 13 days. I am looking at memory
>>> for a few days and noticing that free memory is shrinking
>>> (slowly). Was there any problem with memory leak? :
>>>>> top - 09:46:46 up 13 days, 12:32,  3 users,  load
>>> average: 0.00, 0.04, 0.05
>>>>> Tasks: 159 total,   1 running, 158 sleeping,   0
>>> stopped,   0 zombie
>>>>> Cpu(s):  0.6%us,  0.4%sy,  0.0%ni, 99.0%id,  0.0%wa,
>>>   0.0%hi, 0.0%si, 0.0%st
>>>>> Mem:   4046636k total,  3356340k used,   690296k free,
>>>    357296k buffers
>>>>> Swap:  6094840k total,        0k used,  6094840k free,
>>>   2629280k cached
>>>>> My version: FreeSWITCH Version
>>>>> 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5
>>>>> (1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul
>>> 2012 15:18:27 Z)
>>>>> Thanks!
>>>>>
>>>>> Miha
>>>>>
>>>>>
>> ______________________________________________________________________
>>>>> ___ Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>>> Server
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>>>> rs
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>> ______________________________________________________________________
>>>>> ___ Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>>> Server
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>>>> rs
>>>>> http://www.freeswitch.org
>>>>>
>>>> @Peter thanks for your quick answer:)
>>>>
>>>> I readed jira postes and noticed that this was befor my
>>> git version (as you said).
>>>> If there was leak problem (I do not know if it is, just
>>> noticing that free memory is droping), would FS be stable
>>> for 13 days or would this bug be present sooner (memory
>>> leak)?
>>>> regards,
>>>> Miha
>>>>
>>>>
>>>>
>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>>> Server 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>
>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>>> Server
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>>> Server
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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:5021693632761271914539!
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
@Mitch,

thanks for explanation. So eventually free memory will stop going down?

Otherwise FS is running perfect, I just do not what that anything will 
go wrong while I am not in the office.

p.s.: FS is now up for 14 days. Will memory leak be causing problems any 
sooner or it can influence on memory slowly?

Regards,
Miha


From peter.olsson at visionutveckling.se  Wed Aug  8 10:28:04 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 8 Aug 2012 06:28:04 +0000
Subject: [Freeswitch-users] upadte FS (make current)
Message-ID: <1FFF97C269757C458224B7C895F35F151402EA@cantor.std.visionutv.se>

Since FS only seems to be using 1,4% of the memory it seems fine to me. As mentioned before in this thread, other things like file caching etc. will also use memory, so it's quite normal to see available memory decreasing

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
Skickat: den 8 augusti 2012 08:12
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] upadte FS (make current)

On 8/7/2012 9:51 PM, Peter Olsson wrote:
> Why do you even think FS leaks memory? How much memory does it use?
>
> Anyway, to look for memory leaks, check out this:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Collection_Information_
> With_Valgrind_.28Linux.2FUnix.29
>
> /Peter
>
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Miha 
> [miha at softnet.si]
> Skickat: den 7 augusti 2012 21:22
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] upadte FS (make current)
>
> HI,
>
> just one more information. What is the best way to see what leaks 
> memory and to be sure that FS does not leak it?
>
> If I look at the top command I can see that the free memory is slowly 
> dropping, but from ps aux I can see that FS consums 1.3% of memory 
> (the same).
>
> Thanks!
>
> Miha
>
> Thanks!
>   MIha
>
> On Tue, 07 Aug 2012 12:21:30 +0200
>   Miha <miha at softnet.si> wrote:
>> Hi Peter,
>>
>> fsctl reclaim_mem this will not harm any call and etc?
>>
>> So if freeswitch does not leak memory what should I see in top 
>> (memory will not drop?)?
>>
>> Thanks!
>> Miha
>>
>>
>>
>> On 8/7/2012 11:48 AM, Peter Olsson wrote:
>>> It's always hard to say what is a memory leak and
>> what's not. FS uses memory pools internally, which sometime might 
>> make it look like it leaks memory, but in the end FS will reuse that 
>> memory. You can also run "fsctl reclaim_mem" in the console or 
>> fs_cli, and see if that makes the memory usage drop.
>>> /Peter
>>>
>>> -----Ursprungligt meddelande-----
>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
>> F?r Miha
>>> Skickat: den 7 augusti 2012 11:14
>>> Till: FreeSWITCH Users Help
>>> ?mne: Re: [Freeswitch-users] upadte FS (make current)
>>>
>>> On 8/7/2012 10:54 AM, Peter Olsson wrote:
>>>> Usually it's no problem. However, I always recommend
>> to have a test
>>>> server to run basic tests before going into
>> production. There have
>>>> been a couple of memory leaks fixed, but I think that
>> was before your
>>>> current version - but I don't remember all the commits
>> from my head..
>>>> :)
>>>>
>>>> I would also suggest to read through the git commit
>> log, to get an idea of what has been done.
>>>> /Peter
>>>>
>>>>
>>>> -----Ursprungligt meddelande-----
>>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
>> F?r Miha
>>>> Skickat: den 7 augusti 2012 10:37
>>>> Till: FreeSWITCH Users Help
>>>> ?mne: [Freeswitch-users] upadte FS (make current)
>>>>
>>>> Hi,
>>>>
>>>> I am running on production server FS (runs ok). I
>> would like to upgrade it to latest git.
>>>> Can I expect any problems? I not using any special
>> module (just mod_rad_auth, for cdr mod_xml_cdr).
>>>> Is it better to first test everything on test server
>> and than make upgrade?
>>>> My FS is now up for 13 days. I am looking at memory
>> for a few days and noticing that free memory is shrinking (slowly). 
>> Was there any problem with memory leak? :
>>>> top - 09:46:46 up 13 days, 12:32,  3 users,  load
>> average: 0.00, 0.04, 0.05
>>>> Tasks: 159 total,   1 running, 158 sleeping,   0
>> stopped,   0 zombie
>>>> Cpu(s):  0.6%us,  0.4%sy,  0.0%ni, 99.0%id,  0.0%wa,
>>   0.0%hi, 0.0%si, 0.0%st
>>>> Mem:   4046636k total,  3356340k used,   690296k free,
>>    357296k buffers
>>>> Swap:  6094840k total,        0k used,  6094840k free,
>>   2629280k cached
>>>> My version: FreeSWITCH Version
>>>> 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5
>>>> (1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul
>> 2012 15:18:27 Z)
>>>> Thanks!
>>>>
>>>> Miha
>>>>
>>>>
> ______________________________________________________________________
>>>> ___ Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>>> rs
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>
> ______________________________________________________________________
>>>> ___ Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>>> rs
>>>> http://www.freeswitch.org
>>>>
>>> @Peter thanks for your quick answer:)
>>>
>>> I readed jira postes and noticed that this was befor my
>> git version (as you said).
>>> If there was leak problem (I do not know if it is, just
>> noticing that free memory is droping), would FS be stable for 13 days 
>> or would this bug be present sooner (memory leak)?
>>> regards,
>>> Miha
>>>
>>>
>>>
> ______________________________________________________________________
> ___
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>
> ______________________________________________________________________
> ___
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>> Server
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
>>> http://www.freeswitch.org
>>>
>>
>>
> ______________________________________________________________________
> ___
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>  
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
>> http://www.freeswitch.org
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
> 
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
HI Peter,

I am not thinking that FS memory leaks, I just wont to be shure that my does not leaks:)

Problem is that I am notiching that in top command I can see that memory is slowly shrinking.

For exp in top command yesterday free memory was something about 635348k free

Today:

Mem:   4046636k total,  3511288k used,   535348k free,   367408k buffers
Swap:  6094840k total,        0k used,  6094840k free,  2765076k cached

So every day is a bit litle less.

ps aux:

root      3596  3.9  1.4 457624 58228 ?        S<l  Jul24 827:56 
/usr/local/freeswitch/bin/freeswitch -nc

As this machin is in production and I am going to vacation this week I do not wont to be any problems.

Regards,
Miha

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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Join Us At ClueCon - Aug 7-9, 2012

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!DSPAM:502201c432761188118870!



From miha at softnet.si  Wed Aug  8 10:53:05 2012
From: miha at softnet.si (Miha)
Date: Wed, 08 Aug 2012 08:53:05 +0200
Subject: [Freeswitch-users] upadte FS (make current)
In-Reply-To: <1FFF97C269757C458224B7C895F35F151402EA@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F151402EA@cantor.std.visionutv.se>
Message-ID: <50220CD1.5010002@softnet.si>

On 8/8/2012 8:28 AM, Peter Olsson wrote:
> Since FS only seems to be using 1,4% of the memory it seems fine to me. As mentioned before in this thread, other things like file caching etc. will also use memory, so it's quite normal to see available memory decreasing
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
> Skickat: den 8 augusti 2012 08:12
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] upadte FS (make current)
>
> On 8/7/2012 9:51 PM, Peter Olsson wrote:
>> Why do you even think FS leaks memory? How much memory does it use?
>>
>> Anyway, to look for memory leaks, check out this:
>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Collection_Information_
>> With_Valgrind_.28Linux.2FUnix.29
>>
>> /Peter
>>
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Miha
>> [miha at softnet.si]
>> Skickat: den 7 augusti 2012 21:22
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] upadte FS (make current)
>>
>> HI,
>>
>> just one more information. What is the best way to see what leaks
>> memory and to be sure that FS does not leak it?
>>
>> If I look at the top command I can see that the free memory is slowly
>> dropping, but from ps aux I can see that FS consums 1.3% of memory
>> (the same).
>>
>> Thanks!
>>
>> Miha
>>
>> Thanks!
>>    MIha
>>
>> On Tue, 07 Aug 2012 12:21:30 +0200
>>    Miha <miha at softnet.si> wrote:
>>> Hi Peter,
>>>
>>> fsctl reclaim_mem this will not harm any call and etc?
>>>
>>> So if freeswitch does not leak memory what should I see in top
>>> (memory will not drop?)?
>>>
>>> Thanks!
>>> Miha
>>>
>>>
>>>
>>> On 8/7/2012 11:48 AM, Peter Olsson wrote:
>>>> It's always hard to say what is a memory leak and
>>> what's not. FS uses memory pools internally, which sometime might
>>> make it look like it leaks memory, but in the end FS will reuse that
>>> memory. You can also run "fsctl reclaim_mem" in the console or
>>> fs_cli, and see if that makes the memory usage drop.
>>>> /Peter
>>>>
>>>> -----Ursprungligt meddelande-----
>>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
>>> F?r Miha
>>>> Skickat: den 7 augusti 2012 11:14
>>>> Till: FreeSWITCH Users Help
>>>> ?mne: Re: [Freeswitch-users] upadte FS (make current)
>>>>
>>>> On 8/7/2012 10:54 AM, Peter Olsson wrote:
>>>>> Usually it's no problem. However, I always recommend
>>> to have a test
>>>>> server to run basic tests before going into
>>> production. There have
>>>>> been a couple of memory leaks fixed, but I think that
>>> was before your
>>>>> current version - but I don't remember all the commits
>>> from my head..
>>>>> :)
>>>>>
>>>>> I would also suggest to read through the git commit
>>> log, to get an idea of what has been done.
>>>>> /Peter
>>>>>
>>>>>
>>>>> -----Ursprungligt meddelande-----
>>>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
>>> F?r Miha
>>>>> Skickat: den 7 augusti 2012 10:37
>>>>> Till: FreeSWITCH Users Help
>>>>> ?mne: [Freeswitch-users] upadte FS (make current)
>>>>>
>>>>> Hi,
>>>>>
>>>>> I am running on production server FS (runs ok). I
>>> would like to upgrade it to latest git.
>>>>> Can I expect any problems? I not using any special
>>> module (just mod_rad_auth, for cdr mod_xml_cdr).
>>>>> Is it better to first test everything on test server
>>> and than make upgrade?
>>>>> My FS is now up for 13 days. I am looking at memory
>>> for a few days and noticing that free memory is shrinking (slowly).
>>> Was there any problem with memory leak? :
>>>>> top - 09:46:46 up 13 days, 12:32,  3 users,  load
>>> average: 0.00, 0.04, 0.05
>>>>> Tasks: 159 total,   1 running, 158 sleeping,   0
>>> stopped,   0 zombie
>>>>> Cpu(s):  0.6%us,  0.4%sy,  0.0%ni, 99.0%id,  0.0%wa,
>>>    0.0%hi, 0.0%si, 0.0%st
>>>>> Mem:   4046636k total,  3356340k used,   690296k free,
>>>     357296k buffers
>>>>> Swap:  6094840k total,        0k used,  6094840k free,
>>>    2629280k cached
>>>>> My version: FreeSWITCH Version
>>>>> 1.2.0-rc2+git~20120721T151827Z~b2e28d68b5
>>>>> (1.2.0-rc2; git at commit b2e28d68b5 on Sat, 21 Jul
>>> 2012 15:18:27 Z)
>>>>> Thanks!
>>>>>
>>>>> Miha
>>>>>
>>>>>
>> ______________________________________________________________________
>>>>> ___ Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>>> Server
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>>>> rs
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>> ______________________________________________________________________
>>>>> ___ Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>>> Server
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>>>> rs
>>>>> http://www.freeswitch.org
>>>>>
>>>> @Peter thanks for your quick answer:)
>>>>
>>>> I readed jira postes and noticed that this was befor my
>>> git version (as you said).
>>>> If there was leak problem (I do not know if it is, just
>>> noticing that free memory is droping), would FS be stable for 13 days
>>> or would this bug be present sooner (memory leak)?
>>>> regards,
>>>> Miha
>>>>
>>>>
>>>>
>> ______________________________________________________________________
>> ___
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>>> Server 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>
>> ______________________________________________________________________
>> ___
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication
>>> Server
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>>>> http://www.freeswitch.org
>>>>
>>>
>> ______________________________________________________________________
>> ___
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>>> http://www.freeswitch.org
>> ______________________________________________________________________
>> ___ Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
>>
>>
>>
>> ______________________________________________________________________
>> ___ Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
> HI Peter,
>
> I am not thinking that FS memory leaks, I just wont to be shure that my does not leaks:)
>
> Problem is that I am notiching that in top command I can see that memory is slowly shrinking.
>
> For exp in top command yesterday free memory was something about 635348k free
>
> Today:
>
> Mem:   4046636k total,  3511288k used,   535348k free,   367408k buffers
> Swap:  6094840k total,        0k used,  6094840k free,  2765076k cached
>
> So every day is a bit litle less.
>
> ps aux:
>
> root      3596  3.9  1.4 457624 58228 ?        S<l  Jul24 827:56
> /usr/local/freeswitch/bin/freeswitch -nc
>
> As this machin is in production and I am going to vacation this week I do not wont to be any problems.
>
> Regards,
> Miha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:502201c432761188118870!
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
@Peter,

thank your very much for all your explanation:) Sometime I rather ask 
maybe stupid questions for your but makes me calmer:)

Peter, I see that 1.2 version is out. DO you maybe know when it will be 
available from git? I did upgrade on my test 1.2 rc2 but, after upgrade 
1.2 rc2 is just newer but still 1.2 rc2.

thanks for all your help!

Regards,
Miha


From khuenm at vega.com.vn  Wed Aug  8 05:23:33 2012
From: khuenm at vega.com.vn (Khue Nguyen Minh)
Date: Wed, 8 Aug 2012 08:23:33 +0700
Subject: [Freeswitch-users] Make outbound call
In-Reply-To: <CAAyX+KHGARkzV9t58zjL_xGnRA_5CBq6M-9eF+a2WNwB_fk-Pw@mail.gmail.com>
References: <CAO-yT6hrFqT_hKaqO41GWppsVqHbcSBwkNbMUg1R1Ue03JLAZw@mail.gmail.com>
	<CANJXrA9nxtDTLUuOZRi270Dmy=-s1FowXBXfAtjgw73t7eQ62Q@mail.gmail.com>
	<CAAyX+KHGARkzV9t58zjL_xGnRA_5CBq6M-9eF+a2WNwB_fk-Pw@mail.gmail.com>
Message-ID: <CAO-yT6jwTwhfAHF2PG=ncRWZPh0aRvJn1NTDGt=S==_-1RGC9g@mail.gmail.com>

Hi Ben,

I want use FreeSWITCH in dual mode "Inbound + Outbound". How I can do it?

Brs,
Khue Nguyen.

2012/8/7 Ben Langfeld <ben at langfeld.co.uk>

> You have to use inbound event socket.
>
> Inbound: client connects to freeswitch
> Outbound: freeswitch connects to client
>
> In outbound mode, you only get a connection to the client when you
> have an active call. In order to originate arbitrary calls, you need
> to make a connection from your client in to FreeSWITCH.
>
> Regards,
> Ben Langfeld
> McAfee SiteAdvisor Warning
> This e-mail message contains potentially unsafe links to these sites:
> freeswitchsolutions.com [image: more info...]<https://www.siteadvisor.com/sites/http://www.freeswitchsolutions.com?pip=false&premium=true&client_uid=3485470565&client_ver=3.5.0.227&client_type=IEPlugin&suite=true&aff_id=0&locale=en_us&ui=1&os_ver=5.1.3.0>
>
>
> On 7 August 2012 09:16, Prevail Magid <m.prevail at gmail.com> wrote:
> > You can do it by several way.
> >
> > For example:
> >
> > sendAsyncCommand( channel, "bgapi originate user/1001 777");
> > http://wiki.freeswitch.org/wiki/Mod_commands#originate
> >
> >
> > 2012/8/7 Khue Nguyen Minh <khuenm at vega.com.vn>
> >>
> >> Hi all,
> >>
> >> I am using event socket of FreeSWITCH in "outbound" mode. From my
> >> application, I want request FreeSWITCH make outbound call to a SIP
> client.
> >> How I can do it, please guide me?
> >>
> >> Best regards,
> >> Khue Nguyen.
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ssinyagin at yahoo.com  Wed Aug  8 11:11:56 2012
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Wed, 8 Aug 2012 00:11:56 -0700 (PDT)
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAPcNkddEP9TtOrLu=ikyN_B23QZZfOVBvyqA4Ogx+WDM2V0-uA@mail.gmail.com>
References: <CAPcNkdeC5Yp3ht_4KqYzrHEpVUAYuz_R9gjYi2EDmEepVqLFBQ@mail.gmail.com>
	<20120807182354.68ae72ec@mail.tritonwest.net>
	<CAPcNkddEP9TtOrLu=ikyN_B23QZZfOVBvyqA4Ogx+WDM2V0-uA@mail.gmail.com>
Message-ID: <1344409916.41875.YahooMailNeo@web39305.mail.mud.yahoo.com>

Ken,
care to push tags to Github as well?




>________________________________
> From: Ken Rice <krice at freeswitch.org>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Tuesday, August 7, 2012 8:31 PM
>Subject: Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
> 
>
>yes its there... git pull --tags
>
>
>
>On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>
> 
>>I also checked tags, but didn't see it. Are you sure your branch is tracked upstream?
>>?
>>--Dave
>>
>>
>>>________________________________
>>> From: Ken Rice [mailto:krice at freeswitch.org]
>>>To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
>>>Sent: Tue, 07 Aug 2012 11:15:50 -0700
>>>Subject: Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>>>
>>>
>>>its Tagged, right now, we will be branching it shortly for tracking patches. Please note we had a couple of good users point out some smallish bugs so we're int he process of fixing those! so guys if you find anything wrong please open a jira, if you are at cluecon, find me and let me know the bug number 
>>>
>>>
>>>K
>>>
>>>
>>>On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>>>
>>> 
>>>>Where is the branch so I can checkout local?
>>>>?
>>>>D:\fsnew>git branch -a
>>>>* master
>>>>? remotes/origin/FS-3432
>>>>? remotes/origin/FS-4062
>>>>? remotes/origin/HEAD -> origin/master
>>>>? remotes/origin/dingaling_video
>>>>? remotes/origin/master
>>>>? remotes/origin/smgfs
>>>>? remotes/origin/stable-test/freeswitch-1.2
>>>>? remotes/origin/swk/fs_test
>>>>? remotes/test/master
>>>>
>>>>--Dave
>>>>
>>>>
>>>>>________________________________
>>>>> From: Ken Rice [mailto:krice at freeswitch.org]
>>>>>To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org], freeswitch-dev at lists.freeswitch.org
>>>>>Sent: Tue, 07 Aug 2012 09:03:34 -0700 
>>>>>
>>>>>Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
>>>>>
>>>>>
>>>>>The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0! 
>>>>>
>>>>>
>>>>>Get your copy today at?http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !?
>>>>>
>>>>>
>>>>>Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will continue to fix bugs and security issues. giving you a stable platform for at least one year.
>>>>>
>>>>>
>>>>>Grab it today!
>>>>>
>>>>>
>>>>>The FreeSWITCH Team?
>>>>?
>>>>?
>>>>_________________________________________________________________________
>>>>Professional FreeSWITCH Consulting Services:
>>>>consulting at freeswitch.org
>>>>http://www.freeswitchsolutions.com/
>>>>
>>>>
>>>>/
>>>>
>>>>Official FreeSWITCH Sites
>>>>http://www.freeswitch.org/
>>>>http://wiki.freeswitch.org/
>>>>http://www.cluecon.com/
>>>>
>>>>Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>>FreeSWITCH-users mailing list
>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>http://www.freeswitch.org
>>>>
>>>>
>>>
>>?
>>?
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
>>consulting at freeswitch.org
>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
>>http://www.freeswitch.org
>>http://wiki.freeswitch.org
>>http://www.cluecon.com
>>
>>Join Us At ClueCon - Aug 7-9, 2012
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>Join Us At ClueCon - Aug 7-9, 2012
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.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 telefaks.de  Wed Aug  8 12:07:26 2012
From: lists at telefaks.de (Peter Steinbach)
Date: Wed, 08 Aug 2012 10:07:26 +0200
Subject: [Freeswitch-users] Problem with flex client connecting to
	mod_rtmp
In-Reply-To: <5021399D.2030501@livecall.com>
References: <502064F5.2040702@telefaks.de> <50209901.10107@livecall.com>
	<5020CDB9.3060102@telefaks.de> <5021399D.2030501@livecall.com>
Message-ID: <50221E3E.2060008@telefaks.de>

Thanks Jack,

but as mentioned we are at an earlier stage. So far the Flex client has
not yet conneted to Freeswitch.
We still have a connect problem.

Best regards
Peter

On 08/07/12 17:51, Jack wrote:
> Peter,
>   Here is my rtmp.conf.xml:
>
> <configuration name="rtmp.conf" description="RTMP Endpoint">
>   <profiles>
>     <profile name="default">
>         <settings>
>             <param name="bind-address" value="0.0.0.0:1935" />
>             <param name="context" value="public" />
>             <param name="dialplan" value="XML" />
>             <!-- If this is set to true, no unauthenticated inbound
> calls will be allowed -->
>             <param name="auth-calls" value="true" />
>             <!-- How much time should the clients buffer the media
> stream (miliseconds) -->
>             <param name="buffer-len" value="50" />
>             <!-- Sets the maximum size of outbound RTMP chunks -->
>             <param name="chunksize" value="512" />
>         </settings>
>     </profile>
>   </profiles>
> </configuration>
>
> It is important to specify the context and then make sure you have a
> condition in your corresponding dial plan that will catch your call.
>
> Jack
>
>
>
> On 8/7/2012 1:11 AM, Peter Steinbach wrote:
>> Hello Jack,
>>
>> here are some answers to your your questions:
>>
>>> Make sure you use a fully qualified username not just the extension
>>> 1001 at xxx.xxx.xxx.xxx >the XXX would be the IP of your FreeSwitch Server.
>> I am still in the "connect" state, I am not logging in yet. The webpage
>> tells me "Connecting... " and waits forever
>>
>>> did you configure your rtmp.conf.xml and have a matching context in
>> your dial plan?
>> I am still in the "connect" state
>>
>> In the web page make sure these two vars are set to YOUR Freeswitch
>> server ip: var rtmpIPURL =
>>> "rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx";
>> I did not have a rtmpIP var, so I added it to the code, but no change.
>>
>> In fact as I can see while grepping the network traffic, the Flex app
>> and Freeswicth are negociating their protocol and freeswitch logs a
>> "Sent connect reply", but this is very diferent from what is sent from
>> the conference.freeswitch.org server.
>>
>> Best regards
>> Peter
>>
>>
>> On 08/07/12 06:26, Jack wrote:
>>> Hi Peter,
>>> Make sure you use a fully qualified username not just the extension 
>>> 1001 at xxx.xxx.xxx.xxx
>>> the XXX would be the IP of your FreeSwitch Server.
>>>
>>> did you configure your rtmp.conf.xml and have a matching context in your 
>>> dial plan?
>>>
>>> In the web page make sure these two vars are set to YOUR Freeswitch 
>>> server ip:
>>> var rtmpIPURL = "rtmp://xxx.xxx.xxx.xxx/phone";
>>>   var rtmpIP="xxx.xxx.xxx.xxx";
>>>
>>> jack
>>>
>>> On 8/6/2012 5:44 PM, Peter Steinbach wrote:
>>>> Hello,
>>>>
>>>> today I tried the flex client with the mod_rtmp implementation.
>>>> On a brandnew freeswitch I installed mod_rtmp and copied the flex
>>>> direrectory to a web server and loaded the web page.
>>>>
>>>> However the flex client does not connect:
>>>> Here's my freeswitch log
>>>> 2012-08-07 02:32:03.585671 [NOTICE] mod_rtmp.c:744 New RTMP session
>>>> [f35a192c-0d02-4320-89e9-775a3573ee25]
>>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:702 Sent handshake response
>>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:727 Done with handshake
>>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14
>>>> ts=0 stream_id=0x0] len=342
>>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE
>>>> for connect
>>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1
>>>> stream_id=0x0] len=4
>>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5
>>>> stream_id=0x0] len=4
>>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6
>>>> stream_id=0x0] len=5
>>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4
>>>> stream_id=0x0] len=6
>>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
>>>> stream_id=0x0] len=201
>>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
>>>> stream_id=0x0] len=61
>>>> 2012-08-07 02:32:03.685670 [NOTICE] rtmp_sig.c:121 Sent connect reply
>>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5
>>>> ts=12247364 stream_id=0x0] len=4
>>>> 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
>>>>
>>>> If I change the rtmp_url to 'rtmp://conference.freeswitch.org/phone',
>>>> flex does connect.
>>>>
>>>> But in my freeswitch this fails. I have traced the network traffic and I
>>>> can see that there is information flow between freeswitch and the client
>>>> on connect request.
>>>>
>>>> Somebody has an idea where to look further?
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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


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From admin at blindi.net  Wed Aug  8 12:28:50 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 8 Aug 2012 10:28:50 +0200 (CEST)
Subject: [Freeswitch-users] Lua problem there are only two arguments
 evaluated
In-Reply-To: <CAJUJwtimEQP_r2XMiuK7r5ncidHjM54nCBmdfH1Msa514K5n7A@mail.gmail.com>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<Pine.LNX.4.64.1207311049050.2510@1a-8140.antagus.de>
	<CAJUJwtje2c3i_w9w0aqU0Hwmb0AqifuWS0gMqdAkrsrStBc1jQ@mail.gmail.com>
	<Pine.LNX.4.64.1207311129320.2808@1a-8140.antagus.de>
	<CAJUJwthUm2fuxDet3SfH0tptnLDvB493Nc+NoQN9pZx9ecm+OQ@mail.gmail.com>
	<Pine.LNX.4.64.1208030940330.25711@1a-8140.antagus.de>
	<CAJUJwtimEQP_r2XMiuK7r5ncidHjM54nCBmdfH1Msa514K5n7A@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1208081017060.16625@1a-8140.antagus.de>

Hi SamyGo,

> So, you';ve changed the approach and set the required variable from XML
> dialplan and now you dont need to set that as Argument !?

I have changed my dialplan to:
<action application="set" data="year=${strftime(%Y)}"/>
<action application="set" data="month=${strftime(%m)}"/>
<action application="set" data="day=${strftime(%d)}"/>
<action application="set" data="hours=${strftime(%H)}"/>
<action application="set" data="minutes=${strftime(%M)}"/>
<action application="set" data="seconds=${strftime(%S)}"/>
<action application="set"
data="string=msg_${year}_${month}_${day}_${hours}_${minutes}_
${seconds}_${mailbox}_${caller_id_number}.alaw"/>
I paste the arguments in to lua:
-- parameters: for lua voicemail
-- mailbox, sender, message, messagefile

-- variables
mailbox  = argv[1];
from_box  = argv[2];
msg  = argv[3];
msg_save  = argv[4];



-----------------
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 govoiper at gmail.com  Wed Aug  8 12:38:14 2012
From: govoiper at gmail.com (SamyGo)
Date: Wed, 8 Aug 2012 13:38:14 +0500
Subject: [Freeswitch-users] Lua problem there are only two arguments
	evaluated
In-Reply-To: <Pine.LNX.4.64.1208081017060.16625@1a-8140.antagus.de>
References: <CANNhuhdTnANiyDmQ3G7gCUYBRsEVTC_tRS0XBZf0b42R42UnRQ@mail.gmail.com>
	<50178DB7.7000606@bestnet.kharkov.ua>
	<CAJUJwthC1Uctdey7eVs9Xvb7MeKRP7=sZtG+P_px+3bPN5URmg@mail.gmail.com>
	<Pine.LNX.4.64.1207311049050.2510@1a-8140.antagus.de>
	<CAJUJwtje2c3i_w9w0aqU0Hwmb0AqifuWS0gMqdAkrsrStBc1jQ@mail.gmail.com>
	<Pine.LNX.4.64.1207311129320.2808@1a-8140.antagus.de>
	<CAJUJwthUm2fuxDet3SfH0tptnLDvB493Nc+NoQN9pZx9ecm+OQ@mail.gmail.com>
	<Pine.LNX.4.64.1208030940330.25711@1a-8140.antagus.de>
	<CAJUJwtimEQP_r2XMiuK7r5ncidHjM54nCBmdfH1Msa514K5n7A@mail.gmail.com>
	<Pine.LNX.4.64.1208081017060.16625@1a-8140.antagus.de>
Message-ID: <CAJUJwtg51rd+bR2=ve-TAm1DeSchuJ55Hr25ho80cQ4cFFp0oA@mail.gmail.com>

So this worked for you.?

On Wed, Aug 8, 2012 at 1:28 PM, Thomas Hoellriegel <admin at blindi.net> wrote:

> Hi SamyGo,
>
>
>  So, you';ve changed the approach and set the required variable from XML
>> dialplan and now you dont need to set that as Argument !?
>>
>
> I have changed my dialplan to:
> <action application="set" data="year=${strftime(%Y)}"/>
> <action application="set" data="month=${strftime(%m)}"/>
> <action application="set" data="day=${strftime(%d)}"/>
> <action application="set" data="hours=${strftime(%H)}"/>
> <action application="set" data="minutes=${strftime(%M)}"**/>
> <action application="set" data="seconds=${strftime(%S)}"**/>
> <action application="set"
> data="string=msg_${year}_${**month}_${day}_${hours}_${**minutes}_
> ${seconds}_${mailbox}_${**caller_id_number}.alaw"/>
> I paste the arguments in to lua:
> -- parameters: for lua voicemail
> -- mailbox, sender, message, messagefile
>
> -- variables
>
> mailbox  = argv[1];
> from_box  = argv[2];
> msg  = argv[3];
> msg_save  = argv[4];
>
>
>
> -----------------
>
> 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<http://www.blindi.net/mailman/listinfo/blinde-misc>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.Lappe at qsc.de  Wed Aug  8 12:40:49 2012
From: Adam.Lappe at qsc.de (Lappe, Adam)
Date: Wed, 8 Aug 2012 10:40:49 +0200
Subject: [Freeswitch-users] sendevent to Gateway
In-Reply-To: <CAFGRPVro3EE2ug57k4bkmZHPGbF1EcquQeN_vtQ08dKs-nAxMQ@mail.gmail.com>
References: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EBF@CGNMSG01.oc.qsc.de>
	<F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EC1@CGNMSG01.oc.qsc.de>
	<CAFGRPVro3EE2ug57k4bkmZHPGbF1EcquQeN_vtQ08dKs-nAxMQ@mail.gmail.com>
Message-ID: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EC4@CGNMSG01.oc.qsc.de>

I am trying to send a NOTIFY Message to my Gateway that looks like this:

NOTIFY sip:(address)@(host):(port) SIP/2.0
VIA: SIP/2.0/UDP (address):(port)
From:;tag=1
To:
Call-ID: whatever
CSeq: 1 NOTIFY
Contact:
Event: message-summary
Subscription-State: terminated
Content-Type: application/simple-message-summary
Content-Length: 23

Messages-Waiting: yes


Is it possible to make the FreeSWITCH send such a message to my gateway?

Thanks,
Adam
________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von David Villasmil
Gesendet: Dienstag, 7. August 2012 11:59
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] sendevent to Gateway

Hello,

What EXCATLY do you want to send?

David
On Tue, Aug 7, 2012 at 11:48 AM, Lappe, Adam <Adam.Lappe at qsc.de<mailto:Adam.Lappe at qsc.de>> wrote:
I am still looking for a solution.
Is there no one who has an idea?

Thanks,
Adam

________________________________
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 Lappe, Adam
Gesendet: Freitag, 3. August 2012 11:53
An: 'FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>'
Betreff: [Freeswitch-users] sendevent to Gateway

Hi all,

i am trying to make the freeswitch send a event (or SIP Message) to my gateway.
I can only find examples how to send events to registered endpoint but not to my gateway.

Shouldn?t it be possible to do this via the event socket?

Thanks in advance and best regards,
Adam

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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 asilva at wirelessmundi.com  Wed Aug  8 13:03:41 2012
From: asilva at wirelessmundi.com (Antonio)
Date: Wed, 08 Aug 2012 11:03:41 +0200
Subject: [Freeswitch-users] Problem with flex client connecting to
 mod_rtmp
In-Reply-To: <5020ED57.1040505@telefaks.de>
References: <502064F5.2040702@telefaks.de> <50209901.10107@livecall.com>
	<5020CDB9.3060102@telefaks.de>
	<1344332150.693.7.camel@marces.madrid.commsmundi.com>
	<5020ED57.1040505@telefaks.de>
Message-ID: <1344416621.693.78.camel@marces.madrid.commsmundi.com>

Hi Peter,

Do you have some kind of firewall on your computer? did you try to
disabled it?


Regards,
Ant?nio



On Tue, 2012-08-07 at 12:26 +0200, Peter Steinbach wrote: 

> Hello Antonio,
> 
> I added this:
> > var params  = {
> > allowScriptAccess: 'always'
> > };
> 
> and removed the /phone from the URL. This did not change anything. Log
> is still like this:
> 
> 2012-08-07 11:56:48.105667 [NOTICE] mod_rtmp.c:744 New RTMP session
> [b31edb8a-50d3-4100-9b7f-ab49b6ba06b5]
> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:702 Sent handshake response
> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:727 Done with handshake
> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14
> ts=0 stream_id=0x0] len=331
> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE
> for connect
> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1
> stream_id=0x0] len=4
> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5
> stream_id=0x0] len=4
> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6
> stream_id=0x0] len=5
> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4
> stream_id=0x0] len=6
> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
> stream_id=0x0] len=201
> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
> stream_id=0x0] len=61
> 2012-08-07 11:56:48.205670 [NOTICE] rtmp_sig.c:121 Sent connect reply
> 2012-08-07 11:56:48.215667 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5
> ts=11917881 stream_id=0x0] len=4
> 2012-08-07 11:56:48.215667 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
> 
> Ngrep on the traffic shows the following last packets starting with
> connect: (192.168.178.103 is the PC, 192.168.178.220 is Freeswitch)
> T 192.168.178.103:48179 -> 192.168.178.220:1935 [AP]
> ..............................................................................................K........connect.?..........app.....flashVer...LNX
> 11,2,202,236..swfUrl..,http://my.domain.com/flex/freeswitch.swf..tcUrl...rtmp:/./my.domain.com..fpad....capabilities. at m........audioCodecs.@.........videoCodecs. at o........videoFunction.?.........pageUrl....http://my.domain.com/flex/freeswitch.html#..objectEncoding............
> ##
> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
> ............
> #
> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
> ....
> #
> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
> ............
> #
> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
> ....
> #
> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
> ............
> #
> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
> .....
> #
> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
> ............
> #
> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
> ......
> #
> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
> ............
> ##
> T 192.168.178.103:48179 -> 192.168.178.220:1935 [AP]
> ...9............
> Then traffic stops.
> 
> The packet freeswitch answeres after the connect requests is (from
> wireshark):
> 2.109254    192.168.178.220    192.168.178.220    1935   
> 192.168.178.103    36804    RTMP    Unknown (0x7) | Unknown (0xf3) |
> Unknown (0x21) | Unknown (0x50) | Unknown (0xff)[Packet size limited
> during capture]
> 
> 
> 
> With kind regards
> Peter
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 

Un cordial saludo / Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com
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From admin at blindi.net  Wed Aug  8 13:05:41 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Wed, 8 Aug 2012 11:05:41 +0200 (CEST)
Subject: [Freeswitch-users] Callcenter saying position in queue
In-Reply-To: <501FC899.5030601@gmail.com>
References: <501F86A3.4060904@gmail.com> <501FC899.5030601@gmail.com>
Message-ID: <Pine.LNX.4.64.1208081104350.16625@1a-8140.antagus.de>

Am 06.08.12 um 16:37 schrieb Vbvbrj:

> So I managed to make a script in lua which will count callers position
Can you post your script please?



From paul at cupis.co.uk  Wed Aug  8 13:13:52 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Wed, 8 Aug 2012 10:13:52 +0100
Subject: [Freeswitch-users] upadte FS (make current)
In-Reply-To: <50220CD1.5010002@softnet.si>
References: <1FFF97C269757C458224B7C895F35F151402EA@cantor.std.visionutv.se>
	<50220CD1.5010002@softnet.si>
Message-ID: <20120808091352.GA26167@eagle.cupis.co.uk>

On Wed, Aug 08, 2012 at 08:53:05AM +0200, Miha wrote:
> Peter, I see that 1.2 version is out. DO you maybe know when it will be 
> available from git? I did upgrade on my test 1.2 rc2 but, after upgrade 
> 1.2 rc2 is just newer but still 1.2 rc2.

If you do:

  git pull --tags

and then:

  git tag

You should see there is now a tag for v1.2.0.

I believe there will be "official" instructions on how to use git to track
1.2 available at some point.

Also see:

http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/086608.html
http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/086619.html
http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/086621.html

Regards,




From miha at softnet.si  Wed Aug  8 13:24:40 2012
From: miha at softnet.si (Miha)
Date: Wed, 08 Aug 2012 11:24:40 +0200
Subject: [Freeswitch-users] upadte FS (make current)
In-Reply-To: <20120808091352.GA26167@eagle.cupis.co.uk>
References: <1FFF97C269757C458224B7C895F35F151402EA@cantor.std.visionutv.se>
	<50220CD1.5010002@softnet.si>
	<20120808091352.GA26167@eagle.cupis.co.uk>
Message-ID: <50223058.5090106@softnet.si>

On 8/8/2012 11:13 AM, Paul Cupis wrote:
> On Wed, Aug 08, 2012 at 08:53:05AM +0200, Miha wrote:
>> Peter, I see that 1.2 version is out. DO you maybe know when it will be
>> available from git? I did upgrade on my test 1.2 rc2 but, after upgrade
>> 1.2 rc2 is just newer but still 1.2 rc2.
> If you do:
>
>    git pull --tags
>
> and then:
>
>    git tag
>
> You should see there is now a tag for v1.2.0.
>
> I believe there will be "official" instructions on how to use git to track
> 1.2 available at some point.
>
> Also see:
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/086608.html
> http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/086619.html
> http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/086621.html
>
> Regards,
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
Paul thanks. So i will wait for further instruction:)

BR,

Miha


From lists at telefaks.de  Wed Aug  8 14:07:58 2012
From: lists at telefaks.de (Peter Steinbach)
Date: Wed, 08 Aug 2012 12:07:58 +0200
Subject: [Freeswitch-users] Problem with flex client connecting to
	mod_rtmp
In-Reply-To: <1344416621.693.78.camel@marces.madrid.commsmundi.com>
References: <502064F5.2040702@telefaks.de> <50209901.10107@livecall.com>
	<5020CDB9.3060102@telefaks.de>
	<1344332150.693.7.camel@marces.madrid.commsmundi.com>
	<5020ED57.1040505@telefaks.de>
	<1344416621.693.78.camel@marces.madrid.commsmundi.com>
Message-ID: <50223A7E.2060403@telefaks.de>

Hello Antonio,

it's all on the local network and there is no firewall between. I can
see that messages are exchanged in both directions on port 1935. But the
right connect mesage seems not to be sent by freeswitch.

Best regards
Peter

On 08/08/12 11:03, Antonio wrote:
> Hi Peter,
>
> Do you have some kind of firewall on your computer? did you try to
> disabled it?
>
>
> Regards,
> Ant?nio
>
>
>
> On Tue, 2012-08-07 at 12:26 +0200, Peter Steinbach wrote:
>> Hello Antonio,
>>
>> I added this:
>> > var params  = {
>> > allowScriptAccess: 'always'
>> > };
>>
>> and removed the /phone from the URL. This did not change anything. Log
>> is still like this:
>>
>> 2012-08-07 11:56:48.105667 [NOTICE] mod_rtmp.c:744 New RTMP session
>> [b31edb8a-50d3-4100-9b7f-ab49b6ba06b5]
>> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:702 Sent handshake response
>> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:727 Done with handshake
>> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14
>> ts=0 stream_id=0x0] len=331
>> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE
>> for connect
>> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1
>> stream_id=0x0] len=4
>> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5
>> stream_id=0x0] len=4
>> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6
>> stream_id=0x0] len=5
>> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4
>> stream_id=0x0] len=6
>> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
>> stream_id=0x0] len=201
>> 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
>> stream_id=0x0] len=61
>> 2012-08-07 11:56:48.205670 [NOTICE] rtmp_sig.c:121 Sent connect reply
>> 2012-08-07 11:56:48.215667 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5
>> ts=11917881 stream_id=0x0] len=4
>> 2012-08-07 11:56:48.215667 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
>>
>> Ngrep on the traffic shows the following last packets starting with
>> connect: (192.168.178.103 is the PC, 192.168.178.220 is Freeswitch)
>> T 192.168.178.103:48179 -> 192.168.178.220:1935 [AP]
>> ..............................................................................................K........connect.?..........app.....flashVer...LNX
>> 11,2,202,236..swfUrl..,http://my.domain.com/flex/freeswitch.swf..tcUrl...rtmp:/./my.domain.com..fpad....capabilities. at m........audioCodecs.@.........videoCodecs. at o........videoFunction.?.........pageUrl....http://my.domain.com/flex/freeswitch.html#..objectEncoding............
>> ##
>> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
>> ............
>> #
>> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
>> ....
>> #
>> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
>> ............
>> #
>> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
>> ....
>> #
>> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
>> ............
>> #
>> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
>> .....
>> #
>> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
>> ............
>> #
>> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
>> ......
>> #
>> T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP]
>> ............
>> ##
>> T 192.168.178.103:48179 -> 192.168.178.220:1935 [AP]
>> ...9............
>> Then traffic stops.
>>
>> The packet freeswitch answeres after the connect requests is (from
>> wireshark):
>> 2.109254    192.168.178.220    192.168.178.220    1935   
>> 192.168.178.103    36804    RTMP    Unknown (0x7) | Unknown (0xf3) |
>> Unknown (0x21) | Unknown (0x50) | Unknown (0xff)[Packet size limited
>> during capture]
>>
>>
>>
>> With kind regards
>> Peter
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-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
>
> -- 
>
> Un cordial saludo / Best regards, 
>
>  _________________________
>
> Ant?nio Silva
>
> E-mail:asilva at wirelessmundi.com <mailto:asilva at wirelessmundi.com>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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


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From bdfoster at endigotech.com  Wed Aug  8 15:53:35 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 8 Aug 2012 07:53:35 -0400
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <1344409916.41875.YahooMailNeo@web39305.mail.mud.yahoo.com>
References: <CAPcNkdeC5Yp3ht_4KqYzrHEpVUAYuz_R9gjYi2EDmEepVqLFBQ@mail.gmail.com>
	<20120807182354.68ae72ec@mail.tritonwest.net>
	<CAPcNkddEP9TtOrLu=ikyN_B23QZZfOVBvyqA4Ogx+WDM2V0-uA@mail.gmail.com>
	<1344409916.41875.YahooMailNeo@web39305.mail.mud.yahoo.com>
Message-ID: <CAKV-jFws0meC0=BZvHdLODkGH7OeaeiPoa1UBEOoUg+g5=Ai=g@mail.gmail.com>

FreeSWITCH isn't published on GitHub.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 8, 2012 3:14 AM, "Stanislav Sinyagin" <ssinyagin at yahoo.com> wrote:

> Ken,
> care to push tags to Github as well?
>
>   ------------------------------
> *From:* Ken Rice <krice at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 7, 2012 8:31 PM
> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>
> yes its there... git pull --tags
>
> On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>
> **
> I also checked tags, but didn't see it. Are you sure your branch is
> tracked upstream?
>
> --Dave
>
>  ------------------------------
> *From:* Ken Rice [mailto:krice at freeswitch.org]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> *Sent:* Tue, 07 Aug 2012 11:15:50 -0700
> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>
>
> its Tagged, right now, we will be branching it shortly for tracking
> patches. Please note we had a couple of good users point out some smallish
> bugs so we're int he process of fixing those! so guys if you find anything
> wrong please open a jira, if you are at cluecon, find me and let me know
> the bug number
>
> K
>
> On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>
> **
> Where is the branch so I can checkout local?
>
> D:\fsnew>git branch -a
> * master
>   remotes/origin/FS-3432
>   remotes/origin/FS-4062
>   remotes/origin/HEAD -> origin/master
>   remotes/origin/dingaling_video
>   remotes/origin/master
>   remotes/origin/smgfs
>   remotes/origin/stable-test/freeswitch-1.2
>   remotes/origin/swk/fs_test
>   remotes/test/master
> --Dave
>
>  ------------------------------
> *From:* Ken Rice [mailto:krice at freeswitch.org]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org],
> freeswitch-dev at lists.freeswitch.org
> *Sent:* Tue, 07 Aug 2012 09:03:34 -0700
>
> *Subject:* [Freeswitch-users] Announcing FreeSWITCH 1.2
>
>  The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!
>
> Get your copy today at
> http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !
>
> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
> continue to fix bugs and security issues. giving you a stable platform for
> at least one year.
>
> Grab it today!
>
> The FreeSWITCH Team
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com/
>
> 
> /
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org/
> http://wiki.freeswitch.org/
> http://www.cluecon.com/
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From miha at softnet.si  Wed Aug  8 16:15:40 2012
From: miha at softnet.si (Miha)
Date: Wed, 08 Aug 2012 14:15:40 +0200
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAKV-jFws0meC0=BZvHdLODkGH7OeaeiPoa1UBEOoUg+g5=Ai=g@mail.gmail.com>
References: <CAPcNkdeC5Yp3ht_4KqYzrHEpVUAYuz_R9gjYi2EDmEepVqLFBQ@mail.gmail.com>
	<20120807182354.68ae72ec@mail.tritonwest.net>
	<CAPcNkddEP9TtOrLu=ikyN_B23QZZfOVBvyqA4Ogx+WDM2V0-uA@mail.gmail.com>
	<1344409916.41875.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<CAKV-jFws0meC0=BZvHdLODkGH7OeaeiPoa1UBEOoUg+g5=Ai=g@mail.gmail.com>
Message-ID: <5022586C.5050407@softnet.si>

On 8/8/2012 1:53 PM, Brian Foster wrote:
>
> FreeSWITCH isn't published on GitHub.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
>
> On Aug 8, 2012 3:14 AM, "Stanislav Sinyagin" <ssinyagin at yahoo.com 
> <mailto:ssinyagin at yahoo.com>> wrote:
>
>     Ken,
>     care to push tags to Github as well?
>
>         ------------------------------------------------------------------------
>         *From:* Ken Rice <krice at freeswitch.org
>         <mailto:krice at freeswitch.org>>
>         *To:* FreeSWITCH Users Help
>         <freeswitch-users at lists.freeswitch.org
>         <mailto:freeswitch-users at lists.freeswitch.org>>
>         *Sent:* Tuesday, August 7, 2012 8:31 PM
>         *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>
>         yes its there... git pull --tags
>
>         On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net
>         <mailto:drk at drkngs.net>> wrote:
>
>             I also checked tags, but didn't see it. Are you sure your
>             branch is tracked upstream?
>             --Dave
>
>                 ------------------------------------------------------------------------
>                 *From:* Ken Rice [mailto:krice at freeswitch.org
>                 <mailto:krice at freeswitch.org>]
>                 *To:* FreeSWITCH Users Help
>                 [mailto:freeswitch-users at lists.freeswitch.org
>                 <mailto:freeswitch-users at lists.freeswitch.org>]
>                 *Sent:* Tue, 07 Aug 2012 11:15:50 -0700
>                 *Subject:* Re: [Freeswitch-users] Announcing
>                 FreeSWITCH 1.2
>
>
>                 its Tagged, right now, we will be branching it shortly
>                 for tracking patches. Please note we had a couple of
>                 good users point out some smallish bugs so we're int
>                 he process of fixing those! so guys if you find
>                 anything wrong please open a jira, if you are at
>                 cluecon, find me and let me know the bug number
>
>                 K
>
>                 On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel
>                 <drk at drkngs.net <mailto:drk at drkngs.net>> wrote:
>
>                     Where is the branch so I can checkout local?
>                     D:\fsnew>git branch -a
>                     * master
>                       remotes/origin/FS-3432
>                       remotes/origin/FS-4062
>                       remotes/origin/HEAD -> origin/master
>                     remotes/origin/dingaling_video
>                       remotes/origin/master
>                       remotes/origin/smgfs
>                     remotes/origin/stable-test/freeswitch-1.2
>                     remotes/origin/swk/fs_test
>                       remotes/test/master
>                     --Dave
>
>                         ------------------------------------------------------------------------
>                         *From:* Ken Rice [mailto:krice at freeswitch.org
>                         <mailto:krice at freeswitch.org>]
>                         *To:* FreeSWITCH Users Help
>                         [mailto:freeswitch-users at lists.freeswitch.org
>                         <mailto:freeswitch-users at lists.freeswitch.org>],
>                         freeswitch-dev at lists.freeswitch.org
>                         <mailto:freeswitch-dev at lists.freeswitch.org>
>                         *Sent:* Tue, 07 Aug 2012 09:03:34 -0700
>
>                         *Subject:* [Freeswitch-users] Announcing
>                         FreeSWITCH 1.2
>
>                         The FreeSWITCH Team is Proud to announce
>                         FreeSWITCH 1.2.0!
>
>                         Get your copy today at
>                         http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2
>                         !
>
>                         Going forward FreeSWITCH 1.2.x branch will be
>                         feature stable, but we will continue to fix
>                         bugs and security issues. giving you a stable
>                         platform for at least one year.
>
>                         Grab it today!
>
>                         The FreeSWITCH Team
>
>
>                     _________________________________________________________________________
>                     Professional FreeSWITCH Consulting Services:
>                     consulting at freeswitch.org
>                     <mailto:consulting at freeswitch.org>
>                     http://www.freeswitchsolutions.com/
>
>                     FreeSWITCH-powered IP PBX: The CudaTel
>                     Communication Server
>                     /
>
>                     Official FreeSWITCH Sites
>                     http://www.freeswitch.org/
>                     http://wiki.freeswitch.org/
>                     http://www.cluecon.com/
>
>                     Join Us At ClueCon - Aug 7-9, 2012
>
>                     FreeSWITCH-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/>
>
>
>
>             _________________________________________________________________________
>             Professional FreeSWITCH Consulting Services:
>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>             http://www.freeswitchsolutions.com
>             <http://www.freeswitchsolutions.com/>
>
>             
>              </>
>
>             Official FreeSWITCH Sites
>             http://www.freeswitch.org <http://www.freeswitch.org/>
>             http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>             http://www.cluecon.com <http://www.cluecon.com/>
>
>             Join Us At ClueCon - Aug 7-9, 2012
>
>             FreeSWITCH-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/>
>
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>         <http://www.freeswitchsolutions.com/>
>
>         
>          </>
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org <http://www.freeswitch.org/>
>         http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>         http://www.cluecon.com <http://www.cluecon.com/>
>
>         Join Us At ClueCon - Aug 7-9, 2012
>
>         FreeSWITCH-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/>
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     FreeSWITCH-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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
@Brian, do you maybe know when it will be?

Thanks!

Miha
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From ben at langfeld.co.uk  Wed Aug  8 16:38:10 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Wed, 8 Aug 2012 13:38:10 +0100
Subject: [Freeswitch-users] Make outbound call
In-Reply-To: <CAO-yT6jwTwhfAHF2PG=ncRWZPh0aRvJn1NTDGt=S==_-1RGC9g@mail.gmail.com>
References: <CAO-yT6hrFqT_hKaqO41GWppsVqHbcSBwkNbMUg1R1Ue03JLAZw@mail.gmail.com>
	<CANJXrA9nxtDTLUuOZRi270Dmy=-s1FowXBXfAtjgw73t7eQ62Q@mail.gmail.com>
	<CAAyX+KHGARkzV9t58zjL_xGnRA_5CBq6M-9eF+a2WNwB_fk-Pw@mail.gmail.com>
	<CAO-yT6jwTwhfAHF2PG=ncRWZPh0aRvJn1NTDGt=S==_-1RGC9g@mail.gmail.com>
Message-ID: <CAAyX+KH=iwVtjmNzW66L2zmH6w-YPWR2SomEcMuB5eqDSrjcCA@mail.gmail.com>

The same way you would if you were using either separately :)

Regards,
Ben Langfeld


On 8 August 2012 02:23, Khue Nguyen Minh <khuenm at vega.com.vn> wrote:

> Hi Ben,
>
> I want use FreeSWITCH in dual mode "Inbound + Outbound". How I can do it?
>
> Brs,
> Khue Nguyen.
>
> 2012/8/7 Ben Langfeld <ben at langfeld.co.uk>
>
>> You have to use inbound event socket.
>>
>> Inbound: client connects to freeswitch
>> Outbound: freeswitch connects to client
>>
>> In outbound mode, you only get a connection to the client when you
>> have an active call. In order to originate arbitrary calls, you need
>> to make a connection from your client in to FreeSWITCH.
>>
>> Regards,
>> Ben Langfeld
>>  McAfee SiteAdvisor Warning
>> This e-mail message contains potentially unsafe links to these sites:
>> freeswitchsolutions.com [image: more info...]<https://www.siteadvisor.com/sites/http://www.freeswitchsolutions.com?pip=false&premium=true&client_uid=3485470565&client_ver=3.5.0.227&client_type=IEPlugin&suite=true&aff_id=0&locale=en_us&ui=1&os_ver=5.1.3.0>
>>
>>
>> On 7 August 2012 09:16, Prevail Magid <m.prevail at gmail.com> wrote:
>> > You can do it by several way.
>> >
>> > For example:
>> >
>> > sendAsyncCommand( channel, "bgapi originate user/1001 777");
>> > http://wiki.freeswitch.org/wiki/Mod_commands#originate
>> >
>> >
>> > 2012/8/7 Khue Nguyen Minh <khuenm at vega.com.vn>
>> >>
>> >> Hi all,
>> >>
>> >> I am using event socket of FreeSWITCH in "outbound" mode. From my
>> >> application, I want request FreeSWITCH make outbound call to a SIP
>> client.
>> >> How I can do it, please guide me?
>> >>
>> >> Best regards,
>> >> Khue Nguyen.
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> Join Us At ClueCon - Aug 7-9, 2012
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > Join Us At ClueCon - Aug 7-9, 2012
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ben at langfeld.co.uk  Wed Aug  8 16:39:35 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Wed, 8 Aug 2012 13:39:35 +0100
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAKV-jFws0meC0=BZvHdLODkGH7OeaeiPoa1UBEOoUg+g5=Ai=g@mail.gmail.com>
References: <CAPcNkdeC5Yp3ht_4KqYzrHEpVUAYuz_R9gjYi2EDmEepVqLFBQ@mail.gmail.com>
	<20120807182354.68ae72ec@mail.tritonwest.net>
	<CAPcNkddEP9TtOrLu=ikyN_B23QZZfOVBvyqA4Ogx+WDM2V0-uA@mail.gmail.com>
	<1344409916.41875.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<CAKV-jFws0meC0=BZvHdLODkGH7OeaeiPoa1UBEOoUg+g5=Ai=g@mail.gmail.com>
Message-ID: <CAAyX+KGpuyQ+DQkm9V6dmpuXX6n0j=o9CTr+n_ekM-EOeiBy7A@mail.gmail.com>

What is this then? https://github.com/freeswitch/freeswitch

Regards,
Ben Langfeld


On 8 August 2012 12:53, Brian Foster <bdfoster at endigotech.com> wrote:

> FreeSWITCH isn't published on GitHub.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Aug 8, 2012 3:14 AM, "Stanislav Sinyagin" <ssinyagin at yahoo.com> wrote:
>
>> Ken,
>> care to push tags to Github as well?
>>
>>   ------------------------------
>> *From:* Ken Rice <krice at freeswitch.org>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Tuesday, August 7, 2012 8:31 PM
>> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>>
>> yes its there... git pull --tags
>>
>> On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>>
>> **
>> I also checked tags, but didn't see it. Are you sure your branch is
>> tracked upstream?
>>
>> --Dave
>>
>>  ------------------------------
>> *From:* Ken Rice [mailto:krice at freeswitch.org]
>> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org
>> ]
>> *Sent:* Tue, 07 Aug 2012 11:15:50 -0700
>> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>>
>>
>> its Tagged, right now, we will be branching it shortly for tracking
>> patches. Please note we had a couple of good users point out some smallish
>> bugs so we're int he process of fixing those! so guys if you find anything
>> wrong please open a jira, if you are at cluecon, find me and let me know
>> the bug number
>>
>> K
>>
>> On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>>
>> **
>> Where is the branch so I can checkout local?
>>
>> D:\fsnew>git branch -a
>> * master
>>   remotes/origin/FS-3432
>>   remotes/origin/FS-4062
>>   remotes/origin/HEAD -> origin/master
>>   remotes/origin/dingaling_video
>>   remotes/origin/master
>>   remotes/origin/smgfs
>>   remotes/origin/stable-test/freeswitch-1.2
>>   remotes/origin/swk/fs_test
>>   remotes/test/master
>> --Dave
>>
>>  ------------------------------
>> *From:* Ken Rice [mailto:krice at freeswitch.org]
>> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org],
>> freeswitch-dev at lists.freeswitch.org
>> *Sent:* Tue, 07 Aug 2012 09:03:34 -0700
>>
>> *Subject:* [Freeswitch-users] Announcing FreeSWITCH 1.2
>>
>>  The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!
>>
>> Get your copy today at
>> http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !
>>
>> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
>> continue to fix bugs and security issues. giving you a stable platform for
>> at least one year.
>>
>> Grab it today!
>>
>> The FreeSWITCH Team
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com/
>>
>> 
>> /
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org/
>> http://wiki.freeswitch.org/
>> http://www.cluecon.com/
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Wed Aug  8 17:39:53 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 8 Aug 2012 09:39:53 -0400
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAAyX+KGpuyQ+DQkm9V6dmpuXX6n0j=o9CTr+n_ekM-EOeiBy7A@mail.gmail.com>
References: <CAPcNkdeC5Yp3ht_4KqYzrHEpVUAYuz_R9gjYi2EDmEepVqLFBQ@mail.gmail.com>
	<20120807182354.68ae72ec@mail.tritonwest.net>
	<CAPcNkddEP9TtOrLu=ikyN_B23QZZfOVBvyqA4Ogx+WDM2V0-uA@mail.gmail.com>
	<1344409916.41875.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<CAKV-jFws0meC0=BZvHdLODkGH7OeaeiPoa1UBEOoUg+g5=Ai=g@mail.gmail.com>
	<CAAyX+KGpuyQ+DQkm9V6dmpuXX6n0j=o9CTr+n_ekM-EOeiBy7A@mail.gmail.com>
Message-ID: <CAKV-jFxR99ZFAhn3nK=6HfhKyok0sRR8=a8US1bG2Tzg0X+8WQ@mail.gmail.com>

Ah, looks like they publish to github as well. Afaik the primary repo is
git.freeswitch.org, and that is where I get everything.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 8, 2012 8:41 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:

> What is this then? https://github.com/freeswitch/freeswitch
>
> Regards,
> Ben Langfeld
>
>
> On 8 August 2012 12:53, Brian Foster <bdfoster at endigotech.com> wrote:
>
>> FreeSWITCH isn't published on GitHub.
>>
>> Brian Foster
>> Endigo Computer LLC
>>
>> Sent from a mobile device.
>> On Aug 8, 2012 3:14 AM, "Stanislav Sinyagin" <ssinyagin at yahoo.com> wrote:
>>
>>> Ken,
>>> care to push tags to Github as well?
>>>
>>>   ------------------------------
>>> *From:* Ken Rice <krice at freeswitch.org>
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Sent:* Tuesday, August 7, 2012 8:31 PM
>>> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>>>
>>> yes its there... git pull --tags
>>>
>>> On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>>>
>>> **
>>> I also checked tags, but didn't see it. Are you sure your branch is
>>> tracked upstream?
>>>
>>> --Dave
>>>
>>>  ------------------------------
>>> *From:* Ken Rice [mailto:krice at freeswitch.org]
>>> *To:* FreeSWITCH Users Help [mailto:
>>> freeswitch-users at lists.freeswitch.org]
>>> *Sent:* Tue, 07 Aug 2012 11:15:50 -0700
>>> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>>>
>>>
>>> its Tagged, right now, we will be branching it shortly for tracking
>>> patches. Please note we had a couple of good users point out some smallish
>>> bugs so we're int he process of fixing those! so guys if you find anything
>>> wrong please open a jira, if you are at cluecon, find me and let me know
>>> the bug number
>>>
>>> K
>>>
>>> On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>>>
>>> **
>>> Where is the branch so I can checkout local?
>>>
>>> D:\fsnew>git branch -a
>>> * master
>>>   remotes/origin/FS-3432
>>>   remotes/origin/FS-4062
>>>   remotes/origin/HEAD -> origin/master
>>>   remotes/origin/dingaling_video
>>>   remotes/origin/master
>>>   remotes/origin/smgfs
>>>   remotes/origin/stable-test/freeswitch-1.2
>>>   remotes/origin/swk/fs_test
>>>   remotes/test/master
>>> --Dave
>>>
>>>  ------------------------------
>>> *From:* Ken Rice [mailto:krice at freeswitch.org]
>>> *To:* FreeSWITCH Users Help [mailto:
>>> freeswitch-users at lists.freeswitch.org],
>>> freeswitch-dev at lists.freeswitch.org
>>> *Sent:* Tue, 07 Aug 2012 09:03:34 -0700
>>>
>>> *Subject:* [Freeswitch-users] Announcing FreeSWITCH 1.2
>>>
>>>  The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!
>>>
>>> Get your copy today at
>>> http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !
>>>
>>> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we
>>> will continue to fix bugs and security issues. giving you a stable platform
>>> for at least one year.
>>>
>>> Grab it today!
>>>
>>> The FreeSWITCH Team
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com/
>>>
>>> 
>>> /
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org/
>>> http://wiki.freeswitch.org/
>>> http://www.cluecon.com/
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From eelco at fastmail.nl  Wed Aug  8 16:17:43 2012
From: eelco at fastmail.nl (Eelco)
Date: Wed, 08 Aug 2012 14:17:43 +0200
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
References: <1344419198.28524.140661112182326.206F22F6@webmail.messagingengine.com>
Message-ID: <1344428263.15547.140661112231926.2E36F8FB@webmail.messagingengine.com>

I see a lot of scripts inside .. but what is the correct way to install
this tarball on Debian Squeeze ?

From: Ken Rice [mailto:[1]krice at freeswitch.org]

  To: FreeSWITCH Users Help
  [mailto:[2]freeswitch-users at lists.freeswitch.org],
  [3]freeswitch-dev at lists.freeswitch.org
  Sent: Tue, 07 Aug 2012 09:03:34 -0700

Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2


  The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!



Get your copy today
at http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !



Going forward FreeSWITCH 1.2.x branch will be feature stable, but we
will continue to fix bugs and security issues. giving you a stable
platform for at least one year.



Grab it today!



The FreeSWITCH Team

References

1. mailto:krice at freeswitch.org
2. mailto:freeswitch-users at lists.freeswitch.org
3. mailto:freeswitch-dev at lists.freeswitch.org
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From kris at kriskinc.com  Wed Aug  8 18:54:36 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 8 Aug 2012 10:54:36 -0400
Subject: [Freeswitch-users] Smack a Hacker context?
In-Reply-To: <211601cd7445$26907820$73b16860$@bizfocused.com>
References: <211601cd7445$26907820$73b16860$@bizfocused.com>
Message-ID: <CAKDfjgdVY4Y6GK8qdAAwfiFgSxHt-dnex_p+JTZo49bqNExi4Q@mail.gmail.com>

In 99% of these cases your machine is being hit by a bot (sipvicious,
etc).  Playing media is only going to waste your bandwidth and CPU
resources (chances are they're using a rooted machine).  No human will
ever hear your special message.

Your "adult option" is the best bet.  Drop them, log them, and move on.

On Mon, Aug 6, 2012 at 10:34 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> HI Everyone,
>
>
>
> I didn?t bother to obliterate my servers real identity from my last pastebin
> post, shame on me.  Of course within 24 hours some a$$hole is trying the
> default login ids for Freeswitch.  I am not quite a noob here, but this
> would take me some time to do and I know some of you guys could crank this
> out in minutes.  Of course you all probably have better ideas too.
>
>
>
> I want to allow the users in the sample Freeswitch config to login ? but to
> special context.  In the ?special? context, everything they dial plays a
> recording like ?you?re an ass hangup and try again? or perhaps and ear
> piercing LOUD tone.  Even better would be to dial them every few minutes
> with the same recording or sound.  Best of all, if we can detect that they
> are on a CISCO 5xx phone, I have a special config file to send them!
>
>
>
> Anyone got better ideas?  I would love to hear them.
>
>
>
> I suppose the adult choice would be to gather their external and internal ip
> addresses and report them to their ISP, but that won?t achieve much.  Maybe
> we could build a blacklist and watch for them to connect to the Freeswitch
> conf call.  Then we could all tell them how much we enjoy their efforts.
>
>
>
> Sean
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 krice at freeswitch.org  Wed Aug  8 19:02:29 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 8 Aug 2012 10:02:29 -0500
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAKV-jFxR99ZFAhn3nK=6HfhKyok0sRR8=a8US1bG2Tzg0X+8WQ@mail.gmail.com>
References: <CAPcNkdeC5Yp3ht_4KqYzrHEpVUAYuz_R9gjYi2EDmEepVqLFBQ@mail.gmail.com>
	<20120807182354.68ae72ec@mail.tritonwest.net>
	<CAPcNkddEP9TtOrLu=ikyN_B23QZZfOVBvyqA4Ogx+WDM2V0-uA@mail.gmail.com>
	<1344409916.41875.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<CAKV-jFws0meC0=BZvHdLODkGH7OeaeiPoa1UBEOoUg+g5=Ai=g@mail.gmail.com>
	<CAAyX+KGpuyQ+DQkm9V6dmpuXX6n0j=o9CTr+n_ekM-EOeiBy7A@mail.gmail.com>
	<CAKV-jFxR99ZFAhn3nK=6HfhKyok0sRR8=a8US1bG2Tzg0X+8WQ@mail.gmail.com>
Message-ID: <CAPcNkdcUsMHb3jiMrsKfAPP9ZYOw5bZ7djVdy5Su4tidp0r8jA@mail.gmail.com>

the Primary repo is git.freeswitch.org/freeswitch.git this is the master
repo, however, the github.com/freeswitch/freeswitch is an unofficial
mirror...  updates to primary git repo should be pushed including tags and
branches but we do not support those that wish to use other than official
repos


K


On Wed, Aug 8, 2012 at 8:39 AM, Brian Foster <bdfoster at endigotech.com>wrote:

> Ah, looks like they publish to github as well. Afaik the primary repo is
> git.freeswitch.org, and that is where I get everything.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Aug 8, 2012 8:41 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>
>> What is this then? https://github.com/freeswitch/freeswitch
>>
>> Regards,
>> Ben Langfeld
>>
>>
>> On 8 August 2012 12:53, Brian Foster <bdfoster at endigotech.com> wrote:
>>
>>> FreeSWITCH isn't published on GitHub.
>>>
>>> Brian Foster
>>> Endigo Computer LLC
>>>
>>> Sent from a mobile device.
>>> On Aug 8, 2012 3:14 AM, "Stanislav Sinyagin" <ssinyagin at yahoo.com>
>>> wrote:
>>>
>>>> Ken,
>>>> care to push tags to Github as well?
>>>>
>>>>   ------------------------------
>>>> *From:* Ken Rice <krice at freeswitch.org>
>>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> *Sent:* Tuesday, August 7, 2012 8:31 PM
>>>> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>>>>
>>>> yes its there... git pull --tags
>>>>
>>>> On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>>>>
>>>> **
>>>> I also checked tags, but didn't see it. Are you sure your branch is
>>>> tracked upstream?
>>>>
>>>> --Dave
>>>>
>>>>  ------------------------------
>>>> *From:* Ken Rice [mailto:krice at freeswitch.org]
>>>> *To:* FreeSWITCH Users Help [mailto:
>>>> freeswitch-users at lists.freeswitch.org]
>>>> *Sent:* Tue, 07 Aug 2012 11:15:50 -0700
>>>> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>>>>
>>>>
>>>> its Tagged, right now, we will be branching it shortly for tracking
>>>> patches. Please note we had a couple of good users point out some smallish
>>>> bugs so we're int he process of fixing those! so guys if you find anything
>>>> wrong please open a jira, if you are at cluecon, find me and let me know
>>>> the bug number
>>>>
>>>> K
>>>>
>>>> On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>>>>
>>>> **
>>>> Where is the branch so I can checkout local?
>>>>
>>>> D:\fsnew>git branch -a
>>>> * master
>>>>   remotes/origin/FS-3432
>>>>   remotes/origin/FS-4062
>>>>   remotes/origin/HEAD -> origin/master
>>>>   remotes/origin/dingaling_video
>>>>   remotes/origin/master
>>>>   remotes/origin/smgfs
>>>>   remotes/origin/stable-test/freeswitch-1.2
>>>>   remotes/origin/swk/fs_test
>>>>   remotes/test/master
>>>> --Dave
>>>>
>>>>  ------------------------------
>>>> *From:* Ken Rice [mailto:krice at freeswitch.org]
>>>> *To:* FreeSWITCH Users Help [mailto:
>>>> freeswitch-users at lists.freeswitch.org],
>>>> freeswitch-dev at lists.freeswitch.org
>>>> *Sent:* Tue, 07 Aug 2012 09:03:34 -0700
>>>>
>>>> *Subject:* [Freeswitch-users] Announcing FreeSWITCH 1.2
>>>>
>>>>  The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!
>>>>
>>>> Get your copy today at
>>>> http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !
>>>>
>>>> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we
>>>> will continue to fix bugs and security issues. giving you a stable platform
>>>> for at least one year.
>>>>
>>>> Grab it today!
>>>>
>>>> The FreeSWITCH Team
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com/
>>>>
>>>> 
>>>> /
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org/
>>>> http://wiki.freeswitch.org/
>>>> http://www.cluecon.com/
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Aug  8 19:51:00 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 8 Aug 2012 08:51:00 -0700
Subject: [Freeswitch-users] Smack a Hacker context?
In-Reply-To: <CAKDfjgdVY4Y6GK8qdAAwfiFgSxHt-dnex_p+JTZo49bqNExi4Q@mail.gmail.com>
References: <211601cd7445$26907820$73b16860$@bizfocused.com>
	<CAKDfjgdVY4Y6GK8qdAAwfiFgSxHt-dnex_p+JTZo49bqNExi4Q@mail.gmail.com>
Message-ID: <CAMWiMy8==w5LUWL9D67W_VRw8kK-+5140uG03Zui6Fex_WjdGw@mail.gmail.com>

I acutally used to have a "This number is not in service...... Thank
you from WHHAATTT THHHEEEEEE F************" recording somwhere for
rejected outbound calls.

As krisitian said, just drop it and move on

On Wed, Aug 8, 2012 at 7:54 AM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> In 99% of these cases your machine is being hit by a bot (sipvicious,
> etc).  Playing media is only going to waste your bandwidth and CPU
> resources (chances are they're using a rooted machine).  No human will
> ever hear your special message.
>
> Your "adult option" is the best bet.  Drop them, log them, and move on.
>
> On Mon, Aug 6, 2012 at 10:34 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>> HI Everyone,
>>
>>
>>
>> I didn?t bother to obliterate my servers real identity from my last pastebin
>> post, shame on me.  Of course within 24 hours some a$$hole is trying the
>> default login ids for Freeswitch.  I am not quite a noob here, but this
>> would take me some time to do and I know some of you guys could crank this
>> out in minutes.  Of course you all probably have better ideas too.
>>
>>
>>
>> I want to allow the users in the sample Freeswitch config to login ? but to
>> special context.  In the ?special? context, everything they dial plays a
>> recording like ?you?re an ass hangup and try again? or perhaps and ear
>> piercing LOUD tone.  Even better would be to dial them every few minutes
>> with the same recording or sound.  Best of all, if we can detect that they
>> are on a CISCO 5xx phone, I have a special config file to send them!
>>
>>
>>
>> Anyone got better ideas?  I would love to hear them.
>>
>>
>>
>> I suppose the adult choice would be to gather their external and internal ip
>> addresses and report them to their ISP, but that won?t achieve much.  Maybe
>> we could build a blacklist and watch for them to connect to the Freeswitch
>> conf call.  Then we could all tell them how much we enjoy their efforts.
>>
>>
>>
>> Sean
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From lazyvirus at gmx.com  Wed Aug  8 20:55:18 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Wed, 8 Aug 2012 18:55:18 +0200
Subject: [Freeswitch-users] Smack a Hacker context?
In-Reply-To: <CAMWiMy8==w5LUWL9D67W_VRw8kK-+5140uG03Zui6Fex_WjdGw@mail.gmail.com>
References: <211601cd7445$26907820$73b16860$@bizfocused.com>
	<CAKDfjgdVY4Y6GK8qdAAwfiFgSxHt-dnex_p+JTZo49bqNExi4Q@mail.gmail.com>
	<CAMWiMy8==w5LUWL9D67W_VRw8kK-+5140uG03Zui6Fex_WjdGw@mail.gmail.com>
Message-ID: <20120808185518.4c2e5934@anubis.defcon1>

On Wed, 8 Aug 2012 08:51:00 -0700
curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> I acutally used to have a "This number is not in service......
> Thank you from WHHAATTT THHHEEEEEE F************" recording
> somwhere for rejected outbound calls.

Hmm, how about: "Congratulations, you've reached the FBI VoIP
honeypot bureau.  We'll soon have a little chat that we're 
sure you won't forget... ever" ];-)

JY
-- 
Thom : Maintenant si tu telecharges, ils te coupent ta connexion
       internet, apres ce sera qd tu chantera ss la douche ils
       te couperont l'eau



From jeff at jefflenk.com  Wed Aug  8 21:35:02 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 8 Aug 2012 10:35:02 -0700 (PDT)
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAPcNkdcUsMHb3jiMrsKfAPP9ZYOw5bZ7djVdy5Su4tidp0r8jA@mail.gmail.com>
References: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
	<20120807181233.ee23e2d0@mail.tritonwest.net>
	<CAPcNkdeC5Yp3ht_4KqYzrHEpVUAYuz_R9gjYi2EDmEepVqLFBQ@mail.gmail.com>
	<20120807182354.68ae72ec@mail.tritonwest.net>
	<CAPcNkddEP9TtOrLu=ikyN_B23QZZfOVBvyqA4Ogx+WDM2V0-uA@mail.gmail.com>
	<1344409916.41875.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<CAKV-jFws0meC0=BZvHdLODkGH7OeaeiPoa1UBEOoUg+g5=Ai=g@mail.gmail.com>
	<CAAyX+KGpuyQ+DQkm9V6dmpuXX6n0j=o9CTr+n_ekM-EOeiBy7A@mail.gmail.com>
	<CAKV-jFxR99ZFAhn3nK=6HfhKyok0sRR8=a8US1bG2Tzg0X+8WQ@mail.gmail.com>
	<CAPcNkdcUsMHb3jiMrsKfAPP9ZYOw5bZ7djVdy5Su4tidp0r8jA@mail.gmail.com>
Message-ID: <1344447301918-7581672.post@n2.nabble.com>

Yes the Windows build currently has problems with spandsp. see FS-4504



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Announcing-FreeSWITCH-1-2-tp7581608p7581672.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Wed Aug  8 22:22:02 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 8 Aug 2012 11:22:02 -0700
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <1344428263.15547.140661112231926.2E36F8FB@webmail.messagingengine.com>
References: <1344419198.28524.140661112182326.206F22F6@webmail.messagingengine.com>
	<1344428263.15547.140661112231926.2E36F8FB@webmail.messagingengine.com>
Message-ID: <CAKzWOxWCr5X1547qOwj7Zxvg3JB2KDXrJW9zZxntjb-_e292+g@mail.gmail.com>

On Wed, Aug 8, 2012 at 5:17 AM, Eelco <eelco at fastmail.nl> wrote:

>   I see a lot of scripts inside .. but what is the correct way to install
> this tarball on Debian Squeeze ?
>
>

If this is a simple Linux install (which it sounds like) then start with
these instructions:
http://wiki.freeswitch.org/wiki/Installation_Guide#Debian

-MC
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From avi at avimarcus.net  Wed Aug  8 23:04:50 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 8 Aug 2012 22:04:50 +0300
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAKzWOxWCr5X1547qOwj7Zxvg3JB2KDXrJW9zZxntjb-_e292+g@mail.gmail.com>
References: <1344419198.28524.140661112182326.206F22F6@webmail.messagingengine.com>
	<1344428263.15547.140661112231926.2E36F8FB@webmail.messagingengine.com>
	<CAKzWOxWCr5X1547qOwj7Zxvg3JB2KDXrJW9zZxntjb-_e292+g@mail.gmail.com>
Message-ID: <CALNZuwn+_NpenKCCC1ypS+PDNY9J+EQWYwmy7yw09DgZKk-=Wg@mail.gmail.com>

1) I have some of a Salt-stack install .sls file for freeswitch. I haven't
really used for a new production system, yet, though, so I'm not sure if I
got all the permissions quite right. https://gist.github.com/3297645

2) There's the ubuntu install script for fusionpbx but you can use it for
just FreeSWITCH

-Avi



On Wed, Aug 8, 2012 at 9:22 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 8, 2012 at 5:17 AM, Eelco <eelco at fastmail.nl> wrote:
>
>>   I see a lot of scripts inside .. but what is the correct way to
>> install this tarball on Debian Squeeze ?
>>
>>
>
> If this is a simple Linux install (which it sounds like) then start with
> these instructions:
> http://wiki.freeswitch.org/wiki/Installation_Guide#Debian
>
> -MC
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From leonardo at daitangroup.com  Wed Aug  8 23:17:10 2012
From: leonardo at daitangroup.com (Leonardo)
Date: Wed, 8 Aug 2012 16:17:10 -0300
Subject: [Freeswitch-users] MESSAGE event is being fired twice
Message-ID: <5022BB36.1090009@daitangroup.com>

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From anthony.minessale at gmail.com  Wed Aug  8 23:33:51 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 8 Aug 2012 14:33:51 -0500
Subject: [Freeswitch-users] MESSAGE event is being fired twice
In-Reply-To: <5022BB36.1090009@daitangroup.com>
References: <5022BB36.1090009@daitangroup.com>
Message-ID: <CAKbxfG_OYtS4Yn_cDC3a-k4jeJQGPspj7s08kiEPnLRcdTrzog@mail.gmail.com>

so is it firing once to hit mod sms and fire again when you run the fire app?
are you possibly looking for reply?


On Wed, Aug 8, 2012 at 2:17 PM, Leonardo <leonardo at daitangroup.com> wrote:
> Hi.
> I'm tracking the MESSAGE events on Freeswitch and I noticed that this event
> is being fired twice for each incoming SIP MESSAGE request.
> This is the chatplan I'm using:
>
>     <?xml version="1.0" encoding="utf-8"?>
>     <include>
>       <context name="default">
>
>         <extension name="fire">
>           <condition field="to" expression="^(.*)$">
>             <action application="fire" data=""/>
>           </condition>
>         </extension>
>
>       </context>
>     </include>
>
>
> Does anyone know how to fix it?
>
> Thanks in advance,
> Leo
>
>
>
> --
>
> Leonardo N. S. Pereira, Software Engineer
> T:+55.19.3112-1200 ext. 1283|F:+55.19.3207-1437
> DaitanGroup|www.daitangroup.com|Highly Reliable Outsourcing. Value Added
> Services Worldwide.
> Privileged and confidential. If this message has been received in error,
> please notify sender and delete it immediately.
> Conte?do confidencial. Se esta mensagem foi recebida por engano, favor
> avisar o remetente e apag?-la imediatamente.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 leonardo at daitangroup.com  Thu Aug  9 00:06:22 2012
From: leonardo at daitangroup.com (Leonardo)
Date: Wed, 8 Aug 2012 17:06:22 -0300
Subject: [Freeswitch-users] MESSAGE event is being fired twice
In-Reply-To: <CAKbxfG_OYtS4Yn_cDC3a-k4jeJQGPspj7s08kiEPnLRcdTrzog@mail.gmail.com>
References: <5022BB36.1090009@daitangroup.com>
	<CAKbxfG_OYtS4Yn_cDC3a-k4jeJQGPspj7s08kiEPnLRcdTrzog@mail.gmail.com>
Message-ID: <5022C6BE.6040806@daitangroup.com>

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From hi-tecc at hotmail.com  Thu Aug  9 07:11:53 2012
From: hi-tecc at hotmail.com (DP .)
Date: Wed, 8 Aug 2012 23:11:53 -0400
Subject: [Freeswitch-users] Multiple Skypopen Interface Groups.
Message-ID: <SNT002-W62ED25010D1A5EED84B01EF2CC0@phx.gbl>

Hi List, 
Can we create multiple RR skype interface groups? (with or without the round-robin) or at least choose which interfaces we want in the main RR group?
Thanks, 
Damian 		 	   		  
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From joe.jflemmings at gmail.com  Thu Aug  9 08:10:09 2012
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Wed, 8 Aug 2012 21:10:09 -0700
Subject: [Freeswitch-users] CODEC event in Mod event socket
Message-ID: <CALH5ZrY5__hkX+rtoVfCa46r+q5vEhkwqKVZZTAWZ0Z4TmmEjA@mail.gmail.com>

I?m using Inbound ?Mod event socket? and want to associate the codec event
with a call in the system.

When the event is first fired by FreeSwitch, it has a new Unique-ID and
there is no way to associate it with any other channels.

How can i associate this event so i know what call it belongs to. Please
see sample event bellow on issue. Unique-ID is a new value of
cdee98ff-d786-46be-b058-a29e5eddb0af. I would like to associate this event
with a channel in the system or call leg.


Event-Name: CODEC
Core-UUID: 45e18440-cf99-49ab-bc6e-d85c083312c5
FreeSWITCH-Hostname: freeswitch-64bit-lab
FreeSWITCH-Switchname: freeswitch-64bit-lab
FreeSWITCH-IPv4: 10.10.78.36
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2012-08-08%2020%3A17%3A49
Event-Date-GMT: Thu,%2009%20Aug%202012%2003%3A17%3A49%20GMT
Event-Date-Timestamp: 1344482269907740
Event-Calling-File: switch_core_codec.c
Event-Calling-Function: switch_core_session_set_real_read_codec
Event-Calling-Line-Number: 162
Event-Sequence: 388579
Channel-State: CS_CONSUME_MEDIA
Channel-Call-State: RINGING
Channel-State-Number: 7
Channel-Name:
sofia/external_10.10.78.36/fax_17144301800_8518154256_8518154256_pbxcluster_4500847326%4010.10.78.36
Unique-ID: cdee98ff-d786-46be-b058-a29e5eddb0af
Call-Direction: outbound
Presence-Call-Direction: outbound
Channel-HIT-Dialplan: false
Channel-Call-UUID: cdee98ff-d786-46be-b058-a29e5eddb0af
Answer-State: ringing
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Read-Codec-Bit-Rate: 64000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Channel-Write-Codec-Bit-Rate: 64000
Caller-Direction: outbound
Caller-Caller-ID-Name: 7144301800
Caller-Caller-ID-Number: 7144301800
Caller-Callee-ID-Name: Outbound%20Call
Caller-Callee-ID-Number: 17144301800
Caller-Network-Addr: 10.10.78.36
Caller-Destination-Number:
fax_17144301800_8518154256_8518154256_pbxcluster_4500847326
Caller-Unique-ID: cdee98ff-d786-46be-b058-a29e5eddb0af
Caller-Source: src/switch_ivr_originate.c
Caller-Context: default
Caller-Channel-Name:
sofia/external_10.10.78.36/fax_17144301800_8518154256_8518154256_pbxcluster_4500847326%4010.10.78.36
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1344482263028410
Caller-Channel-Created-Time: 1344482263028410
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
channel-read-codec-name: PCMU
channel-read-codec-rate: 8000
channel-read-codec-bit-rate: 64000
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From gmaruzz at gmail.com  Thu Aug  9 08:59:49 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 8 Aug 2012 23:59:49 -0500
Subject: [Freeswitch-users] Multiple Skypopen Interface Groups.
In-Reply-To: <SNT002-W62ED25010D1A5EED84B01EF2CC0@phx.gbl>
References: <SNT002-W62ED25010D1A5EED84B01EF2CC0@phx.gbl>
Message-ID: <CALXCt0qnYVWAqASYUGpJOMyMT3Yzo+12Jwv-GdW2TPwL9EKixg@mail.gmail.com>

no, not at this moment.

Open a Jira issue with a feature request.

You can also (additionally) offer a bounty.

-giovanni

On 8/8/12, DP . <hi-tecc at hotmail.com> wrote:
> Hi List,
> Can we create multiple RR skype interface groups? (with or without the
> round-robin) or at least choose which interfaces we want in the main RR
> group?
> Thanks,
> Damian 		 	   		


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From miha at softnet.si  Thu Aug  9 10:14:52 2012
From: miha at softnet.si (Miha)
Date: Thu, 09 Aug 2012 08:14:52 +0200
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAPcNkdcUsMHb3jiMrsKfAPP9ZYOw5bZ7djVdy5Su4tidp0r8jA@mail.gmail.com>
References: <CAPcNkdeC5Yp3ht_4KqYzrHEpVUAYuz_R9gjYi2EDmEepVqLFBQ@mail.gmail.com>
	<20120807182354.68ae72ec@mail.tritonwest.net>
	<CAPcNkddEP9TtOrLu=ikyN_B23QZZfOVBvyqA4Ogx+WDM2V0-uA@mail.gmail.com>
	<1344409916.41875.YahooMailNeo@web39305.mail.mud.yahoo.com>
	<CAKV-jFws0meC0=BZvHdLODkGH7OeaeiPoa1UBEOoUg+g5=Ai=g@mail.gmail.com>
	<CAAyX+KGpuyQ+DQkm9V6dmpuXX6n0j=o9CTr+n_ekM-EOeiBy7A@mail.gmail.com>
	<CAKV-jFxR99ZFAhn3nK=6HfhKyok0sRR8=a8US1bG2Tzg0X+8WQ@mail.gmail.com>
	<CAPcNkdcUsMHb3jiMrsKfAPP9ZYOw5bZ7djVdy5Su4tidp0r8jA@mail.gmail.com>
Message-ID: <5023555C.3020306@softnet.si>

Hi,

so my current version is 1.2 rc2. After make current I will have 1.2? I 
try it on test server but still FreeSWITCH Version 
1.2.0-rc2+git~20120809T022906Z~3389f32363 (1.2.0-rc2; git at commit 
3389f32363 on Thu, 09 Aug 2012 02:29:06 Z)


Thanks!

Miha

On 8/8/2012 5:02 PM, Ken Rice wrote:
> the Primary repo is git.freeswitch.org/freeswitch.git 
> <http://git.freeswitch.org/freeswitch.git> this is the master repo, 
> however, the github.com/freeswitch/freeswitch 
> <http://github.com/freeswitch/freeswitch> is an unofficial mirror... 
>  updates to primary git repo should be pushed including tags and 
> branches but we do not support those that wish to use other than 
> official repos
>
>
> K
>
>
> On Wed, Aug 8, 2012 at 8:39 AM, Brian Foster <bdfoster at endigotech.com 
> <mailto:bdfoster at endigotech.com>> wrote:
>
>     Ah, looks like they publish to github as well. Afaik the primary
>     repo is git.freeswitch.org <http://git.freeswitch.org>, and that
>     is where I get everything.
>
>     Brian Foster
>     Endigo Computer LLC
>
>     Sent from a mobile device.
>
>     On Aug 8, 2012 8:41 AM, "Ben Langfeld" <ben at langfeld.co.uk
>     <mailto:ben at langfeld.co.uk>> wrote:
>
>         What is this then? https://github.com/freeswitch/freeswitch
>
>         Regards,
>         Ben Langfeld
>
>
>         On 8 August 2012 12:53, Brian Foster <bdfoster at endigotech.com
>         <mailto:bdfoster at endigotech.com>> wrote:
>
>             FreeSWITCH isn't published on GitHub.
>
>             Brian Foster
>             Endigo Computer LLC
>
>             Sent from a mobile device.
>
>             On Aug 8, 2012 3:14 AM, "Stanislav Sinyagin"
>             <ssinyagin at yahoo.com <mailto:ssinyagin at yahoo.com>> wrote:
>
>                 Ken,
>                 care to push tags to Github as well?
>
>                     ------------------------------------------------------------------------
>                     *From:* Ken Rice <krice at freeswitch.org
>                     <mailto:krice at freeswitch.org>>
>                     *To:* FreeSWITCH Users Help
>                     <freeswitch-users at lists.freeswitch.org
>                     <mailto:freeswitch-users at lists.freeswitch.org>>
>                     *Sent:* Tuesday, August 7, 2012 8:31 PM
>                     *Subject:* Re: [Freeswitch-users] Announcing
>                     FreeSWITCH 1.2
>
>                     yes its there... git pull --tags
>
>                     On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel
>                     <drk at drkngs.net <mailto:drk at drkngs.net>> wrote:
>
>                         I also checked tags, but didn't see it. Are
>                         you sure your branch is tracked upstream?
>                         --Dave
>
>                             ------------------------------------------------------------------------
>                             *From:* Ken Rice
>                             [mailto:krice at freeswitch.org
>                             <mailto:krice at freeswitch.org>]
>                             *To:* FreeSWITCH Users Help
>                             [mailto:freeswitch-users at lists.freeswitch.org
>                             <mailto:freeswitch-users at lists.freeswitch.org>]
>                             *Sent:* Tue, 07 Aug 2012 11:15:50 -0700
>                             *Subject:* Re: [Freeswitch-users]
>                             Announcing FreeSWITCH 1.2
>
>
>                             its Tagged, right now, we will be
>                             branching it shortly for tracking patches.
>                             Please note we had a couple of good users
>                             point out some smallish bugs so we're int
>                             he process of fixing those! so guys if you
>                             find anything wrong please open a jira, if
>                             you are at cluecon, find me and let me
>                             know the bug number
>
>                             K
>
>                             On Tue, Aug 7, 2012 at 1:12 PM, Dave R.
>                             Kompel <drk at drkngs.net
>                             <mailto:drk at drkngs.net>> wrote:
>
>                                 Where is the branch so I can checkout
>                                 local?
>                                 D:\fsnew>git branch -a
>                                 * master
>                                 remotes/origin/FS-3432
>                                 remotes/origin/FS-4062
>                                 remotes/origin/HEAD -> origin/master
>                                 remotes/origin/dingaling_video
>                                 remotes/origin/master
>                                 remotes/origin/smgfs
>                                 remotes/origin/stable-test/freeswitch-1.2
>                                 remotes/origin/swk/fs_test
>                                 remotes/test/master
>                                 --Dave
>
>                                     ------------------------------------------------------------------------
>                                     *From:* Ken Rice
>                                     [mailto:krice at freeswitch.org
>                                     <mailto:krice at freeswitch.org>]
>                                     *To:* FreeSWITCH Users Help
>                                     [mailto:freeswitch-users at lists.freeswitch.org
>                                     <mailto:freeswitch-users at lists.freeswitch.org>],
>                                     freeswitch-dev at lists.freeswitch.org <mailto:freeswitch-dev at lists.freeswitch.org>
>                                     *Sent:* Tue, 07 Aug 2012 09:03:34
>                                     -0700
>
>                                     *Subject:* [Freeswitch-users]
>                                     Announcing FreeSWITCH 1.2
>
>                                     The FreeSWITCH Team is Proud to
>                                     announce FreeSWITCH 1.2.0!
>
>                                     Get your copy today at
>                                     http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2
>                                     !
>
>                                     Going forward FreeSWITCH 1.2.x
>                                     branch will be feature stable, but
>                                     we will continue to fix bugs and
>                                     security issues. giving you a
>                                     stable platform for at least one year.
>
>                                     Grab it today!
>
>                                     The FreeSWITCH Team
>
>
>                                 _________________________________________________________________________
>                                 Professional FreeSWITCH Consulting
>                                 Services:
>                                 consulting at freeswitch.org
>                                 <mailto:consulting at freeswitch.org>
>                                 http://www.freeswitchsolutions.com/
>
>                                 FreeSWITCH-powered IP PBX: The CudaTel
>                                 Communication Server
>                                 /
>
>                                 Official FreeSWITCH Sites
>                                 http://www.freeswitch.org/
>                                 http://wiki.freeswitch.org/
>                                 http://www.cluecon.com/
>
>                                 Join Us At ClueCon - Aug 7-9, 2012
>
>                                 FreeSWITCH-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/>
>
>
>
>                         _________________________________________________________________________
>                         Professional FreeSWITCH Consulting Services:
>                         consulting at freeswitch.org
>                         <mailto:consulting at freeswitch.org>
>                         http://www.freeswitchsolutions.com
>                         <http://www.freeswitchsolutions.com/>
>
>                         FreeSWITCH-powered IP PBX: The CudaTel
>                         Communication Server
>                          </>
>
>                         Official FreeSWITCH Sites
>                         http://www.freeswitch.org
>                         <http://www.freeswitch.org/>
>                         http://wiki.freeswitch.org
>                         <http://wiki.freeswitch.org/>
>                         http://www.cluecon.com <http://www.cluecon.com/>
>
>                         Join Us At ClueCon - Aug 7-9, 2012
>
>                         FreeSWITCH-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/>
>
>
>
>                     _________________________________________________________________________
>                     Professional FreeSWITCH Consulting Services:
>                     consulting at freeswitch.org
>                     <mailto:consulting at freeswitch.org>
>                     http://www.freeswitchsolutions.com
>                     <http://www.freeswitchsolutions.com/>
>
>                     FreeSWITCH-powered IP PBX: The CudaTel
>                     Communication Server
>                      </>
>
>                     Official FreeSWITCH Sites
>                     http://www.freeswitch.org <http://www.freeswitch.org/>
>                     http://wiki.freeswitch.org
>                     <http://wiki.freeswitch.org/>
>                     http://www.cluecon.com <http://www.cluecon.com/>
>
>                     Join Us At ClueCon - Aug 7-9, 2012
>
>                     FreeSWITCH-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/>
>
>
>
>                 _________________________________________________________________________
>                 Professional FreeSWITCH Consulting Services:
>                 consulting at freeswitch.org
>                 <mailto:consulting at freeswitch.org>
>                 http://www.freeswitchsolutions.com
>
>                 FreeSWITCH-powered IP PBX: The CudaTel Communication
>                 Server
>                 
>
>                 Official FreeSWITCH Sites
>                 http://www.freeswitch.org
>                 http://wiki.freeswitch.org
>                 http://www.cluecon.com
>
>                 Join Us At ClueCon - Aug 7-9, 2012
>
>                 FreeSWITCH-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
>
>
>             _________________________________________________________________________
>             Professional FreeSWITCH Consulting Services:
>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>             http://www.freeswitchsolutions.com
>
>             
>             
>
>             Official FreeSWITCH Sites
>             http://www.freeswitch.org
>             http://wiki.freeswitch.org
>             http://www.cluecon.com
>
>             Join Us At ClueCon - Aug 7-9, 2012
>
>             FreeSWITCH-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
>
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>
>         Join Us At ClueCon - Aug 7-9, 2012
>
>         FreeSWITCH-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
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     FreeSWITCH-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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Aug  9 10:36:53 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 9 Aug 2012 06:36:53 +0000
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
Message-ID: <1FFF97C269757C458224B7C895F35F15141094@cantor.std.visionutv.se>

If you change to branch "v1.2.stable", the version will be 1.2.0.

The core devs have not yet sent out any official instructions, but I guess that if you want to continue to follow the main development (with sometimes unstable code), just continue to do as you do today. If you want to stick to the 1.2 version, and only get bugfixes etc. for that version (but no new features), change to the branch v1.2.stable.

I think there will be more official instructions soon, but this is that way I think it's meant to work.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
Skickat: den 9 augusti 2012 08:15
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Announcing FreeSWITCH 1.2

Hi,

so my current version is 1.2 rc2. After make current I will have 1.2? I try it on test server but still FreeSWITCH Version 1.2.0-rc2+git~20120809T022906Z~3389f32363 (1.2.0-rc2; git at commit 3389f32363 on Thu, 09 Aug 2012 02:29:06 Z)


Thanks!

Miha

On 8/8/2012 5:02 PM, Ken Rice wrote:
the Primary repo is git.freeswitch.org/freeswitch.git<http://git.freeswitch.org/freeswitch.git> this is the master repo, however, the github.com/freeswitch/freeswitch<http://github.com/freeswitch/freeswitch> is an unofficial mirror...  updates to primary git repo should be pushed including tags and branches but we do not support those that wish to use other than official repos


K

On Wed, Aug 8, 2012 at 8:39 AM, Brian Foster <bdfoster at endigotech.com<mailto:bdfoster at endigotech.com>> wrote:

Ah, looks like they publish to github as well. Afaik the primary repo is git.freeswitch.org<http://git.freeswitch.org>, and that is where I get everything.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 8, 2012 8:41 AM, "Ben Langfeld" <ben at langfeld.co.uk<mailto:ben at langfeld.co.uk>> wrote:
What is this then? https://github.com/freeswitch/freeswitch

Regards,
Ben Langfeld

On 8 August 2012 12:53, Brian Foster <bdfoster at endigotech.com<mailto:bdfoster at endigotech.com>> wrote:

FreeSWITCH isn't published on GitHub.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 8, 2012 3:14 AM, "Stanislav Sinyagin" <ssinyagin at yahoo.com<mailto:ssinyagin at yahoo.com>> wrote:
Ken,
care to push tags to Github as well?

________________________________
From: Ken Rice <krice at freeswitch.org<mailto:krice at freeswitch.org>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Tuesday, August 7, 2012 8:31 PM
Subject: Re: [Freeswitch-users] Announcing FreeSWITCH 1.2

yes its there... git pull --tags

On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net<mailto:drk at drkngs.net>> wrote:
I also checked tags, but didn't see it. Are you sure your branch is tracked upstream?

--Dave

________________________________
From: Ken Rice [mailto:krice at freeswitch.org<mailto:krice at freeswitch.org>]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>]
Sent: Tue, 07 Aug 2012 11:15:50 -0700
Subject: Re: [Freeswitch-users] Announcing FreeSWITCH 1.2


its Tagged, right now, we will be branching it shortly for tracking patches. Please note we had a couple of good users point out some smallish bugs so we're int he process of fixing those! so guys if you find anything wrong please open a jira, if you are at cluecon, find me and let me know the bug number

K
On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net<mailto:drk at drkngs.net>> wrote:
Where is the branch so I can checkout local?

D:\fsnew>git branch -a
* master
  remotes/origin/FS-3432
  remotes/origin/FS-4062
  remotes/origin/HEAD -> origin/master
  remotes/origin/dingaling_video
  remotes/origin/master
  remotes/origin/smgfs
  remotes/origin/stable-test/freeswitch-1.2
  remotes/origin/swk/fs_test
  remotes/test/master
--Dave

________________________________
From: Ken Rice [mailto:krice at freeswitch.org<mailto:krice at freeswitch.org>]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>], freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>
Sent: Tue, 07 Aug 2012 09:03:34 -0700

Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!

Get your copy today at http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !

Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will continue to fix bugs and security issues. giving you a stable platform for at least one year.

Grab it today!

The FreeSWITCH Team



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com/


/

Official FreeSWITCH Sites
http://www.freeswitch.org/
http://wiki.freeswitch.org/
http://www.cluecon.com/

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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/>




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>


</>

Official FreeSWITCH Sites
http://www.freeswitch.org<http://www.freeswitch.org/>
http://wiki.freeswitch.org<http://wiki.freeswitch.org/>
http://www.cluecon.com<http://www.cluecon.com/>

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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/>


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>


</>

Official FreeSWITCH Sites
http://www.freeswitch.org<http://www.freeswitch.org/>
http://wiki.freeswitch.org<http://wiki.freeswitch.org/>
http://www.cluecon.com<http://www.cluecon.com/>

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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/>


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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





_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org<mailto:consulting at freeswitch.org>

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com



Join Us At ClueCon - Aug 7-9, 2012



FreeSWITCH-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:502354a432765721010175!
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From wingcomm at hotmail.com  Thu Aug  9 09:09:57 2012
From: wingcomm at hotmail.com (R W)
Date: Thu, 9 Aug 2012 01:09:57 -0400
Subject: [Freeswitch-users] TLS on FreeSwitch not Working
Message-ID: <BAY156-W11F916CCDCFC6B2A6B3E2DD4CC0@phx.gbl>


Hi All,
I cannot seem to get TLS running on the sofia "internal" profile. Any assistance would be appreciated.
I'm running FreeSWITCH Version 1.2.0-rc2+git~20120808T025758Z~9ac586adc8 (1.2.0-rc2; git at commit 9ac586adc8 on Wed, 08 Aug 2012 02:57:58 Z) on Ubuntu 12.04 LTS.
When I set internal_ssl_enable=true, and reload the sofia internal profile, I get the "usual" error:
2012-08-09 00:34:14.174431 [ERR] sofia.c:2289 Error Creating SIP UA for profile: internal
<!-- SIP Profile: Internal --><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=true"/><X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>
I verified that the OpenSSL development libraries were installed (Ubuntu package libssl-dev) and looked for references to ssl in the output from the compilation process and saw this:








"checking for openssl... yes
checking openssl_CFLAGS...  
checking openssl_LIBS... -lssl -lcrypto  
  adding "-DHAVE_OPENSSL" to SWITCH_AM_CFLAGS"
...
"checking OpenSSL options with pkg-config... found








checking for gdi32... no
checking for CRYPTO_lock in -lcrypto... yes
checking for SSL_connect in -lssl... yes
checking openssl/x509.h usability... yes
checking openssl/x509.h presence... yes
checking for openssl/x509.h... yes
checking openssl/rsa.h usability... yes
checking openssl/rsa.h presence... yes
checking for openssl/rsa.h... yes
checking openssl/crypto.h usability... yes
checking openssl/crypto.h presence... yes
checking for openssl/crypto.h... yes
checking openssl/pem.h usability... yes
checking openssl/pem.h presence... yes
checking for openssl/pem.h... yes
checking openssl/ssl.h usability... yes
checking openssl/ssl.h presence... yes
checking for openssl/ssl.h... yes
checking openssl/err.h usability... yes
checking openssl/err.h presence... yes
checking for openssl/err.h... yes
checking openssl/pkcs12.h usability... yes
checking openssl/pkcs12.h presence... yes
checking for openssl/pkcs12.h... yes
checking for ENGINE_init... yes
checking openssl/engine.h usability... yes
checking openssl/engine.h presence... yes







checking for openssl/engine.h... yes
checking for ENGINE_load_builtin_engines... yes
checking for RAND_status... yes
checking for RAND_screen... no
checking for RAND_egd... yes
checking for CRYPTO_cleanup_all_ex_data... yes
checking for "/dev/urandom"... yes
checking CA cert bundle install path... ${prefix}/share/curl/curl-ca-bundle.crt
checking for inflateEnd in -lz... yes"
Is there anything else I should be checking. Does freeswitch send logs anywhere other than ../freeswitch/log/ ?


 		 	   		  
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From miha at softnet.si  Thu Aug  9 11:11:21 2012
From: miha at softnet.si (Miha)
Date: Thu, 09 Aug 2012 09:11:21 +0200
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <1FFF97C269757C458224B7C895F35F15141094@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F15141094@cantor.std.visionutv.se>
Message-ID: <50236299.4070403@softnet.si>

On 8/9/2012 8:36 AM, Peter Olsson wrote:
>
> If you change to branch "v1.2.stable", the version will be 1.2.0.
>
> The core devs have not yet sent out any official instructions, but I 
> guess that if you want to continue to follow the main development 
> (with sometimes unstable code), just continue to do as you do today. 
> If you want to stick to the 1.2 version, and only get bugfixes etc. 
> for that version (but no new features), change to the branch v1.2.stable.
>
> I think there will be more official instructions soon, but this is 
> that way I think it's meant to work.
>
> /Peter
>
> *Fr?n:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *F?r *Miha
> *Skickat:* den 9 augusti 2012 08:15
> *Till:* FreeSWITCH Users Help
> *?mne:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>
> Hi,
>
> so my current version is 1.2 rc2. After make current I will have 1.2? 
> I try it on test server but still FreeSWITCH Version 
> 1.2.0-rc2+git~20120809T022906Z~3389f32363 (1.2.0-rc2; git at commit 
> 3389f32363 on Thu, 09 Aug 2012 02:29:06 Z)
>
>
> Thanks!
>
> Miha
>
> On 8/8/2012 5:02 PM, Ken Rice wrote:
>
>     the Primary repo is git.freeswitch.org/freeswitch.git
>     <http://git.freeswitch.org/freeswitch.git> this is the master
>     repo, however, the github.com/freeswitch/freeswitch
>     <http://github.com/freeswitch/freeswitch> is an unofficial
>     mirror...  updates to primary git repo should be pushed including
>     tags and branches but we do not support those that wish to use
>     other than official repos
>
>     K
>
>     On Wed, Aug 8, 2012 at 8:39 AM, Brian Foster
>     <bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>> wrote:
>
>     Ah, looks like they publish to github as well. Afaik the primary
>     repo is git.freeswitch.org <http://git.freeswitch.org>, and that
>     is where I get everything.
>
>     Brian Foster
>     Endigo Computer LLC
>
>     Sent from a mobile device.
>
>     On Aug 8, 2012 8:41 AM, "Ben Langfeld" <ben at langfeld.co.uk
>     <mailto:ben at langfeld.co.uk>> wrote:
>
>     What is this then? https://github.com/freeswitch/freeswitch
>
>
>     Regards,
>     Ben Langfeld
>
>     On 8 August 2012 12:53, Brian Foster <bdfoster at endigotech.com
>     <mailto:bdfoster at endigotech.com>> wrote:
>
>     FreeSWITCH isn't published on GitHub.
>
>     Brian Foster
>     Endigo Computer LLC
>
>     Sent from a mobile device.
>
>     On Aug 8, 2012 3:14 AM, "Stanislav Sinyagin" <ssinyagin at yahoo.com
>     <mailto:ssinyagin at yahoo.com>> wrote:
>
>     Ken,
>
>     care to push tags to Github as well?
>
>         ------------------------------------------------------------------------
>
>         *From:*Ken Rice <krice at freeswitch.org
>         <mailto:krice at freeswitch.org>>
>         *To:* FreeSWITCH Users Help
>         <freeswitch-users at lists.freeswitch.org
>         <mailto:freeswitch-users at lists.freeswitch.org>>
>         *Sent:* Tuesday, August 7, 2012 8:31 PM
>         *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>
>         yes its there... git pull --tags
>
>         On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net
>         <mailto:drk at drkngs.net>> wrote:
>
>         I also checked tags, but didn't see it. Are you sure your
>         branch is tracked upstream?
>
>         --Dave
>
>             ------------------------------------------------------------------------
>
>             *From:* Ken Rice [mailto:krice at freeswitch.org
>             <mailto:krice at freeswitch.org>]
>             *To:* FreeSWITCH Users Help
>             [mailto:freeswitch-users at lists.freeswitch.org
>             <mailto:freeswitch-users at lists.freeswitch.org>]
>
>             *Sent:* Tue, 07 Aug 2012 11:15:50 -0700
>             *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>
>
>
>             its Tagged, right now, we will be branching it shortly for
>             tracking patches. Please note we had a couple of good
>             users point out some smallish bugs so we're int he process
>             of fixing those! so guys if you find anything wrong please
>             open a jira, if you are at cluecon, find me and let me
>             know the bug number
>
>             K
>
>             On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel
>             <drk at drkngs.net <mailto:drk at drkngs.net>> wrote:
>
>             Where is the branch so I can checkout local?
>
>             D:\fsnew>git branch -a
>             * master
>             remotes/origin/FS-3432
>             remotes/origin/FS-4062
>             remotes/origin/HEAD -> origin/master
>             remotes/origin/dingaling_video
>             remotes/origin/master
>             remotes/origin/smgfs
>             remotes/origin/stable-test/freeswitch-1.2
>             remotes/origin/swk/fs_test
>             remotes/test/master
>
>             --Dave
>
>                 ------------------------------------------------------------------------
>
>                 *From:* Ken Rice [mailto:krice at freeswitch.org
>                 <mailto:krice at freeswitch.org>]
>                 *To:* FreeSWITCH Users Help
>                 [mailto:freeswitch-users at lists.freeswitch.org
>                 <mailto:freeswitch-users at lists.freeswitch.org>],
>                 freeswitch-dev at lists.freeswitch.org
>                 <mailto:freeswitch-dev at lists.freeswitch.org>
>                 *Sent:* Tue, 07 Aug 2012 09:03:34 -0700
>
>
>                 *Subject:* [Freeswitch-users] Announcing FreeSWITCH 1.2
>
>                 The FreeSWITCH Team is Proud to announce FreeSWITCH
>                 1.2.0!
>
>                 Get your copy today at
>                 http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !
>
>                 Going forward FreeSWITCH 1.2.x branch will be feature
>                 stable, but we will continue to fix bugs and security
>                 issues. giving you a stable platform for at least one
>                 year.
>
>                 Grab it today!
>
>                 The FreeSWITCH Team
>
>
>             _________________________________________________________________________
>             Professional FreeSWITCH Consulting Services:
>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>             http://www.freeswitchsolutions.com/
>
>             
>             /
>
>             Official FreeSWITCH Sites
>             http://www.freeswitch.org/
>             http://wiki.freeswitch.org/
>             http://www.cluecon.com/
>
>             Join Us At ClueCon - Aug 7-9, 2012
>
>             FreeSWITCH-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/>
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>         <http://www.freeswitchsolutions.com/>
>
>         
>          </>
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org <http://www.freeswitch.org/>
>         http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>         http://www.cluecon.com <http://www.cluecon.com/>
>
>         Join Us At ClueCon - Aug 7-9, 2012
>
>         FreeSWITCH-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/>
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>         <http://www.freeswitchsolutions.com/>
>
>         
>          </>
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org <http://www.freeswitch.org/>
>         http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>         http://www.cluecon.com <http://www.cluecon.com/>
>
>         Join Us At ClueCon - Aug 7-9, 2012
>
>         FreeSWITCH-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/>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     FreeSWITCH-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
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     FreeSWITCH-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
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     FreeSWITCH-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
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     FreeSWITCH-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
>
>
>
>
>     _________________________________________________________________________
>
>     Professional FreeSWITCH Consulting Services:
>
>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>
>     http://www.freeswitchsolutions.com
>
>       
>
>     
>
>     
>
>       
>
>     Official FreeSWITCH Sites
>
>     http://www.freeswitch.org
>
>     http://wiki.freeswitch.org
>
>     http://www.cluecon.com
>
>       
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>       
>
>     FreeSWITCH-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:502354a432765721010175!
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
@Peter,

thanks for all your explanation.

Br,
Miha
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From miha at softnet.si  Thu Aug  9 12:39:25 2012
From: miha at softnet.si (Miha)
Date: Thu, 09 Aug 2012 10:39:25 +0200
Subject: [Freeswitch-users] FS 1.2 make problems
Message-ID: <5023773D.4040007@softnet.si>

HI,

today I tried to install FS 1.2 on test server but after make I get this:

make  all-am
make[3]: Entering directory `/usr/local/src/freeswitch/libs/curl/src'
make[3]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
make[2]: Entering directory `/usr/local/src/freeswitch/libs/curl'
make[2]: Nothing to be done for `all-am'.
make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
make[1]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
cat /usr/local/src/freeswitch/src/include/switch_cpp.h | perl /usr/local/src/freeswitch/build/strip.pl > /usr/local/src/freeswitch/src/include/switch_swigable_cpp.h
make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
cc1: warnings being treated as errors
In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:114,
                  from ./src/include/private/switch_core_pvt.h:35,
                  from src/switch.c:48:
/usr/local/src/freeswitch/libs/spandsp/src/spandsp/t30.h:381: warning: ISO C restricts enumerator values to range of ???int???
make[1]: *** [freeswitch-switch.o] Error 1
make: *** [all] Error 2
You have mail in /var/spool/mail/root

Thanks!

Miha





From peter.olsson at visionutveckling.se  Thu Aug  9 12:55:40 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 9 Aug 2012 08:55:40 +0000
Subject: [Freeswitch-users] FS 1.2 make problems
Message-ID: <1FFF97C269757C458224B7C895F35F15141324@cantor.std.visionutv.se>

Add to Jira please...

FYI: It was fixed in commit  3818cae, but then reverted again in commit  65ffaa8.

Submit the issue to Jira, and set Steve Underwood as the assignee, he's the author of libspandsp.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
Skickat: den 9 augusti 2012 10:39
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] FS 1.2 make problems

HI,

today I tried to install FS 1.2 on test server but after make I get this:

make  all-am
make[3]: Entering directory `/usr/local/src/freeswitch/libs/curl/src'
make[3]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
make[2]: Entering directory `/usr/local/src/freeswitch/libs/curl'
make[2]: Nothing to be done for `all-am'.
make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
make[1]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
cat /usr/local/src/freeswitch/src/include/switch_cpp.h | perl /usr/local/src/freeswitch/build/strip.pl > /usr/local/src/freeswitch/src/include/switch_swigable_cpp.h
make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
cc1: warnings being treated as errors
In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:114,
                  from ./src/include/private/switch_core_pvt.h:35,
                  from src/switch.c:48:
/usr/local/src/freeswitch/libs/spandsp/src/spandsp/t30.h:381: warning: ISO C restricts enumerator values to range of ???int???
make[1]: *** [freeswitch-switch.o] Error 1
make: *** [all] Error 2
You have mail in /var/spool/mail/root

Thanks!

Miha




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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Join Us At ClueCon - Aug 7-9, 2012

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!DSPAM:5023758a32769079885987!


From miha at softnet.si  Thu Aug  9 13:18:00 2012
From: miha at softnet.si (Miha)
Date: Thu, 09 Aug 2012 11:18:00 +0200
Subject: [Freeswitch-users] FS 1.2 make problems
In-Reply-To: <1FFF97C269757C458224B7C895F35F15141324@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F15141324@cantor.std.visionutv.se>
Message-ID: <50238048.8040703@softnet.si>

On 8/9/2012 10:55 AM, Peter Olsson wrote:
> Add to Jira please...
>
> FYI: It was fixed in commit  3818cae, but then reverted again in commit  65ffaa8.
>
> Submit the issue to Jira, and set Steve Underwood as the assignee, he's the author of libspandsp.
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
> Skickat: den 9 augusti 2012 10:39
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] FS 1.2 make problems
>
> HI,
>
> today I tried to install FS 1.2 on test server but after make I get this:
>
> make  all-am
> make[3]: Entering directory `/usr/local/src/freeswitch/libs/curl/src'
> make[3]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
> make[2]: Entering directory `/usr/local/src/freeswitch/libs/curl'
> make[2]: Nothing to be done for `all-am'.
> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
> cat /usr/local/src/freeswitch/src/include/switch_cpp.h | perl /usr/local/src/freeswitch/build/strip.pl > /usr/local/src/freeswitch/src/include/switch_swigable_cpp.h
> make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
> cc1: warnings being treated as errors
> In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:114,
>                    from ./src/include/private/switch_core_pvt.h:35,
>                    from src/switch.c:48:
> /usr/local/src/freeswitch/libs/spandsp/src/spandsp/t30.h:381: warning: ISO C restricts enumerator values to range of ???int???
> make[1]: *** [freeswitch-switch.o] Error 1
> make: *** [all] Error 2
> You have mail in /var/spool/mail/root
>
> Thanks!
>
> Miha
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:5023758a32769079885987!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
I did Peter.

Thanks for info.

Regards,
Miha



From godson.g at gmail.com  Thu Aug  9 15:27:58 2012
From: godson.g at gmail.com (Godson Gera)
Date: Thu, 9 Aug 2012 16:57:58 +0530
Subject: [Freeswitch-users] CODEC event in Mod event socket
In-Reply-To: <CALH5ZrY5__hkX+rtoVfCa46r+q5vEhkwqKVZZTAWZ0Z4TmmEjA@mail.gmail.com>
References: <CALH5ZrY5__hkX+rtoVfCa46r+q5vEhkwqKVZZTAWZ0Z4TmmEjA@mail.gmail.com>
Message-ID: <CAPTXd0o8sRHzRQ7noQTn+5p54vf4ttzTyrFk0BDq5FNLZwMRsA@mail.gmail.com>

You must already be having a channel in the FS with that unique id. Are you
originating these calls from event socket ?

On Thu, Aug 9, 2012 at 9:40 AM, Joe Flemmings <joe.jflemmings at gmail.com>wrote:

> I?m using Inbound ?Mod event socket? and want to associate the codec event
> with a call in the system.
>
> When the event is first fired by FreeSwitch, it has a new Unique-ID and
> there is no way to associate it with any other channels.
>
> How can i associate this event so i know what call it belongs to. Please
> see sample event bellow on issue. Unique-ID is a new value of
> cdee98ff-d786-46be-b058-a29e5eddb0af. I would like to associate this event
> with a channel in the system or call leg.
>
>
> Event-Name: CODEC
> Core-UUID: 45e18440-cf99-49ab-bc6e-d85c083312c5
> FreeSWITCH-Hostname: freeswitch-64bit-lab
> FreeSWITCH-Switchname: freeswitch-64bit-lab
> FreeSWITCH-IPv4: 10.10.78.36
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2012-08-08%2020%3A17%3A49
> Event-Date-GMT: Thu,%2009%20Aug%202012%2003%3A17%3A49%20GMT
> Event-Date-Timestamp: 1344482269907740
> Event-Calling-File: switch_core_codec.c
> Event-Calling-Function: switch_core_session_set_real_read_codec
> Event-Calling-Line-Number: 162
> Event-Sequence: 388579
> Channel-State: CS_CONSUME_MEDIA
> Channel-Call-State: RINGING
> Channel-State-Number: 7
> Channel-Name: sofia/external_10.10.78.36/fax_17144301800
> _8518154256_8518154256_pbxcluster_4500847326%4010.10.78.36
> Unique-ID: cdee98ff-d786-46be-b058-a29e5eddb0af
> Call-Direction: outbound
> Presence-Call-Direction: outbound
> Channel-HIT-Dialplan: false
> Channel-Call-UUID: cdee98ff-d786-46be-b058-a29e5eddb0af
> Answer-State: ringing
> Channel-Read-Codec-Name: PCMU
> Channel-Read-Codec-Rate: 8000
> Channel-Read-Codec-Bit-Rate: 64000
> Channel-Write-Codec-Name: PCMU
> Channel-Write-Codec-Rate: 8000
> Channel-Write-Codec-Bit-Rate: 64000
> Caller-Direction: outbound
> Caller-Caller-ID-Name: 7144301800
> Caller-Caller-ID-Number: 7144301800
> Caller-Callee-ID-Name: Outbound%20Call
> Caller-Callee-ID-Number: 17144301800
> Caller-Network-Addr: 10.10.78.36
> Caller-Destination-Number: fax_17144301800
> _8518154256_8518154256_pbxcluster_4500847326
> Caller-Unique-ID: cdee98ff-d786-46be-b058-a29e5eddb0af
> Caller-Source: src/switch_ivr_originate.c
> Caller-Context: default
> Caller-Channel-Name: sofia/external_10.10.78.36/fax_17144301800
> _8518154256_8518154256_pbxcluster_4500847326%4010.10.78.36
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1344482263028410
> Caller-Channel-Created-Time: 1344482263028410
> 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
> channel-read-codec-name: PCMU
> channel-read-codec-rate: 8000
> channel-read-codec-bit-rate: 64000
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Thanks & Regards,
Godson Gera
FreeSWITCH Consultant <http://godson.in>
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From steveu at coppice.org  Thu Aug  9 15:42:15 2012
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 09 Aug 2012 19:42:15 +0800
Subject: [Freeswitch-users] FS 1.2 make problems
In-Reply-To: <1FFF97C269757C458224B7C895F35F15141324@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F15141324@cantor.std.visionutv.se>
Message-ID: <5023A217.5080407@coppice.org>

If 65ffaa8 causes the reported failure to compile, someone is using a 
broken compiler. Which platform is this?

Steve

On 08/09/2012 04:55 PM, Peter Olsson wrote:
> Add to Jira please...
>
> FYI: It was fixed in commit  3818cae, but then reverted again in commit  65ffaa8.
>
> Submit the issue to Jira, and set Steve Underwood as the assignee, he's the author of libspandsp.
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
> Skickat: den 9 augusti 2012 10:39
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] FS 1.2 make problems
>
> HI,
>
> today I tried to install FS 1.2 on test server but after make I get this:
>
> make  all-am
> make[3]: Entering directory `/usr/local/src/freeswitch/libs/curl/src'
> make[3]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
> make[2]: Entering directory `/usr/local/src/freeswitch/libs/curl'
> make[2]: Nothing to be done for `all-am'.
> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
> cat /usr/local/src/freeswitch/src/include/switch_cpp.h | perl /usr/local/src/freeswitch/build/strip.pl > /usr/local/src/freeswitch/src/include/switch_swigable_cpp.h
> make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
> cc1: warnings being treated as errors
> In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:114,
>                    from ./src/include/private/switch_core_pvt.h:35,
>                    from src/switch.c:48:
> /usr/local/src/freeswitch/libs/spandsp/src/spandsp/t30.h:381: warning: ISO C restricts enumerator values to range of ???int???
> make[1]: *** [freeswitch-switch.o] Error 1
> make: *** [all] Error 2
> You have mail in /var/spool/mail/root
>
> Thanks!
>
> Miha
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:5023758a32769079885987!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From miha at softnet.si  Thu Aug  9 15:56:17 2012
From: miha at softnet.si (Miha)
Date: Thu, 09 Aug 2012 13:56:17 +0200
Subject: [Freeswitch-users] FS 1.2 make problems
In-Reply-To: <5023A217.5080407@coppice.org>
References: <1FFF97C269757C458224B7C895F35F15141324@cantor.std.visionutv.se>
	<5023A217.5080407@coppice.org>
Message-ID: <5023A561.2040503@softnet.si>

On 8/9/2012 1:42 PM, Steve Underwood wrote:
> If 65ffaa8 causes the reported failure to compile, someone is using a
> broken compiler. Which platform is this?
>
> Steve
>
> On 08/09/2012 04:55 PM, Peter Olsson wrote:
>> Add to Jira please...
>>
>> FYI: It was fixed in commit  3818cae, but then reverted again in commit  65ffaa8.
>>
>> Submit the issue to Jira, and set Steve Underwood as the assignee, he's the author of libspandsp.
>>
>> /Peter
>>
>>
>> -----Ursprungligt meddelande-----
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
>> Skickat: den 9 augusti 2012 10:39
>> Till: FreeSWITCH Users Help
>> ?mne: [Freeswitch-users] FS 1.2 make problems
>>
>> HI,
>>
>> today I tried to install FS 1.2 on test server but after make I get this:
>>
>> make  all-am
>> make[3]: Entering directory `/usr/local/src/freeswitch/libs/curl/src'
>> make[3]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
>> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
>> make[2]: Entering directory `/usr/local/src/freeswitch/libs/curl'
>> make[2]: Nothing to be done for `all-am'.
>> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
>> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
>> cat /usr/local/src/freeswitch/src/include/switch_cpp.h | perl /usr/local/src/freeswitch/build/strip.pl > /usr/local/src/freeswitch/src/include/switch_swigable_cpp.h
>> make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
>> cc1: warnings being treated as errors
>> In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:114,
>>                     from ./src/include/private/switch_core_pvt.h:35,
>>                     from src/switch.c:48:
>> /usr/local/src/freeswitch/libs/spandsp/src/spandsp/t30.h:381: warning: ISO C restricts enumerator values to range of ???int???
>> make[1]: *** [freeswitch-switch.o] Error 1
>> make: *** [all] Error 2
>> You have mail in /var/spool/mail/root
>>
>> Thanks!
>>
>> Miha
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>  
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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:5023758a32769079885987!
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
Hi @Steve,

linux version is:

CentOS release 5.8 (Final)

FS 1.2 rc2 was normally compiled.

Regards,
Miha





From peter.olsson at visionutveckling.se  Thu Aug  9 15:58:47 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 9 Aug 2012 11:58:47 +0000
Subject: [Freeswitch-users] FS 1.2 make problems
Message-ID: <1FFF97C269757C458224B7C895F35F151414C0@cantor.std.visionutv.se>

Steve, it's reported at: FS-4509

The original change (commit 3818cae) mentioned that this problem was on CentOS. I haven't tried myself though.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Steve Underwood
Skickat: den 9 augusti 2012 13:42
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] FS 1.2 make problems

If 65ffaa8 causes the reported failure to compile, someone is using a 
broken compiler. Which platform is this?

Steve

On 08/09/2012 04:55 PM, Peter Olsson wrote:
> Add to Jira please...
>
> FYI: It was fixed in commit  3818cae, but then reverted again in commit  65ffaa8.
>
> Submit the issue to Jira, and set Steve Underwood as the assignee, he's the author of libspandsp.
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
> Skickat: den 9 augusti 2012 10:39
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] FS 1.2 make problems
>
> HI,
>
> today I tried to install FS 1.2 on test server but after make I get this:
>
> make  all-am
> make[3]: Entering directory `/usr/local/src/freeswitch/libs/curl/src'
> make[3]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
> make[2]: Entering directory `/usr/local/src/freeswitch/libs/curl'
> make[2]: Nothing to be done for `all-am'.
> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
> cat /usr/local/src/freeswitch/src/include/switch_cpp.h | perl /usr/local/src/freeswitch/build/strip.pl > /usr/local/src/freeswitch/src/include/switch_swigable_cpp.h
> make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
> cc1: warnings being treated as errors
> In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:114,
>                    from ./src/include/private/switch_core_pvt.h:35,
>                    from src/switch.c:48:
> /usr/local/src/freeswitch/libs/spandsp/src/spandsp/t30.h:381: warning: ISO C restricts enumerator values to range of ???int???
> make[1]: *** [freeswitch-switch.o] Error 1
> make: *** [all] Error 2
> You have mail in /var/spool/mail/root
>
> Thanks!
>
> Miha
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> 
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at 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:5023a00532762097718087!


From david.villasmil.work at gmail.com  Thu Aug  9 16:07:16 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 9 Aug 2012 14:07:16 +0200
Subject: [Freeswitch-users] sendevent to Gateway
In-Reply-To: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EC4@CGNMSG01.oc.qsc.de>
References: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EBF@CGNMSG01.oc.qsc.de>
	<F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EC1@CGNMSG01.oc.qsc.de>
	<CAFGRPVro3EE2ug57k4bkmZHPGbF1EcquQeN_vtQ08dKs-nAxMQ@mail.gmail.com>
	<F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EC4@CGNMSG01.oc.qsc.de>
Message-ID: <CAFGRPVqby1X4JxtEFgKeGEre6yRugsvsOFND4JSoUbfMJSDRKQ@mail.gmail.com>

Have you tried:

http://wiki.freeswitch.org/wiki/Event_Socket#sendevent

Example of sendevent with a message body, the length of the body is
specified by content-length:

sendevent NOTIFY
profile: internal
content-type: application/simple-message-summary
event-string: check-sync
user: 1005
host: 192.168.10.4
content-length: 2

OK

Another example with a notify:


sendevent NOTIFY
profile: internal
content-type: application/simple-message-summary
event-string: check-sync
user: 1005
host: 99.157.44.194
content-length: 2

OK

Results in a packet like this:


NOTIFY sip:1005 at 99.157.44.203 SIP/2.0
Via: SIP/2.0/UDP 99.157.44.194;rport;branch=z9hG4bKpH2DtBDcDtg0N
Max-Forwards: 70
From: <sip:1005 at 99.157.44.194>;tag=Dy3c6Q1y15v5S
To: <sip:1005 at 99.157.44.194>
Call-ID: 129d1446-0063-122c-15aa-001a923f6a0f
CSeq: 104766492 NOTIFY
Contact: <sip:mod_sofia at 99.157.44.194:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9578:9586
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: 100rel, timer, precondition, path, replaces
Event: check-sync
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary
Subscription-State: terminated;timeout
Content-Type: application/simple-message-summary
Content-Length: 2

OK


On Wed, Aug 8, 2012 at 10:40 AM, Lappe, Adam <Adam.Lappe at qsc.de> wrote:

>  I am trying to send a NOTIFY Message to my Gateway that looks like this:*
> ***
>
> ** **
>
> NOTIFY sip:(address)@(host):(port) SIP/2.0****
>
> VIA: SIP/2.0/UDP (address):(port)****
>
> From:;tag=1****
>
> To:****
>
> Call-ID: whatever****
>
> CSeq: 1 NOTIFY****
>
> Contact:****
>
> Event: message-summary****
>
> Subscription-State: terminated****
>
> Content-Type: application/simple-message-summary****
>
> Content-Length: 23****
>
> ** **
>
> Messages-Waiting: yes****
>
> ** **
>
> ** **
>
> Is it possible to make the FreeSWITCH send such a message to my gateway?**
> **
>
> ** **
>
> Thanks,****
>
> Adam****
>  ------------------------------
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *David
> Villasmil
> *Gesendet:* Dienstag, 7. August 2012 11:59
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] sendevent to Gateway****
>
> ** **
>
> Hello,****
>
> ** **
>
> What EXCATLY do you want to send?****
>
> ** **
>
> David****
>
> On Tue, Aug 7, 2012 at 11:48 AM, Lappe, Adam <Adam.Lappe at qsc.de> wrote:***
> *
>
> I am still looking for a solution.****
>
> Is there no one who has an idea?****
>
>  ****
>
> Thanks,****
>
> Adam****
>
>  ****
>  ------------------------------
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Lappe,
> Adam
> *Gesendet:* Freitag, 3. August 2012 11:53
> *An:* 'FreeSWITCH-users at lists.freeswitch.org'
> *Betreff:* [Freeswitch-users] sendevent to Gateway****
>
>  ****
>
> Hi all,****
>
>  ****
>
> i am trying to make the freeswitch send a event (or SIP Message) to my
> gateway.****
>
> I can only find examples how to send events to registered endpoint but not
> to my gateway.****
>
>  ****
>
> Shouldn?t it be possible to do this via the event socket?****
>
>  ****
>
> Thanks in advance and best regards,****
>
> Adam****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.villasmil.work at gmail.com  Thu Aug  9 16:08:43 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 9 Aug 2012 14:08:43 +0200
Subject: [Freeswitch-users] Make outbound call
In-Reply-To: <CAAyX+KH=iwVtjmNzW66L2zmH6w-YPWR2SomEcMuB5eqDSrjcCA@mail.gmail.com>
References: <CAO-yT6hrFqT_hKaqO41GWppsVqHbcSBwkNbMUg1R1Ue03JLAZw@mail.gmail.com>
	<CANJXrA9nxtDTLUuOZRi270Dmy=-s1FowXBXfAtjgw73t7eQ62Q@mail.gmail.com>
	<CAAyX+KHGARkzV9t58zjL_xGnRA_5CBq6M-9eF+a2WNwB_fk-Pw@mail.gmail.com>
	<CAO-yT6jwTwhfAHF2PG=ncRWZPh0aRvJn1NTDGt=S==_-1RGC9g@mail.gmail.com>
	<CAAyX+KH=iwVtjmNzW66L2zmH6w-YPWR2SomEcMuB5eqDSrjcCA@mail.gmail.com>
Message-ID: <CAFGRPVokCVQT7a6hb4xhKmFkQcziW=Zm0jT-9Rvw8jDwzbi-oQ@mail.gmail.com>

Meaning (I'm guessing) setup your dialplan to connect outbound to your
script, which when you started it up connected "inbound" to FS...



On Wed, Aug 8, 2012 at 2:38 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> The same way you would if you were using either separately :)
>
> Regards,
> Ben Langfeld
>
>
>
> On 8 August 2012 02:23, Khue Nguyen Minh <khuenm at vega.com.vn> wrote:
>
>> Hi Ben,
>>
>> I want use FreeSWITCH in dual mode "Inbound + Outbound". How I can do it?
>>
>> Brs,
>> Khue Nguyen.
>>
>> 2012/8/7 Ben Langfeld <ben at langfeld.co.uk>
>>
>>> You have to use inbound event socket.
>>>
>>> Inbound: client connects to freeswitch
>>> Outbound: freeswitch connects to client
>>>
>>> In outbound mode, you only get a connection to the client when you
>>> have an active call. In order to originate arbitrary calls, you need
>>> to make a connection from your client in to FreeSWITCH.
>>>
>>> Regards,
>>> Ben Langfeld
>>>  McAfee SiteAdvisor Warning
>>> This e-mail message contains potentially unsafe links to these sites:
>>> freeswitchsolutions.com [image: more info...]<https://www.siteadvisor.com/sites/http://www.freeswitchsolutions.com?pip=false&premium=true&client_uid=3485470565&client_ver=3.5.0.227&client_type=IEPlugin&suite=true&aff_id=0&locale=en_us&ui=1&os_ver=5.1.3.0>
>>>
>>>
>>> On 7 August 2012 09:16, Prevail Magid <m.prevail at gmail.com> wrote:
>>> > You can do it by several way.
>>> >
>>> > For example:
>>> >
>>> > sendAsyncCommand( channel, "bgapi originate user/1001 777");
>>> > http://wiki.freeswitch.org/wiki/Mod_commands#originate
>>> >
>>> >
>>> > 2012/8/7 Khue Nguyen Minh <khuenm at vega.com.vn>
>>> >>
>>> >> Hi all,
>>> >>
>>> >> I am using event socket of FreeSWITCH in "outbound" mode. From my
>>> >> application, I want request FreeSWITCH make outbound call to a SIP
>>> client.
>>> >> How I can do it, please guide me?
>>> >>
>>> >> Best regards,
>>> >> Khue Nguyen.
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> 
>>> >> 
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://wiki.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> Join Us At ClueCon - Aug 7-9, 2012
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > Join Us At ClueCon - Aug 7-9, 2012
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From miha at softnet.si  Thu Aug  9 16:12:18 2012
From: miha at softnet.si (Miha)
Date: Thu, 09 Aug 2012 14:12:18 +0200
Subject: [Freeswitch-users] FS 1.2 make problems
In-Reply-To: <1FFF97C269757C458224B7C895F35F151414C0@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F151414C0@cantor.std.visionutv.se>
Message-ID: <5023A922.3070503@softnet.si>

Steve, I tried to assigned ticket to you as Peter suggested but I guess 
it did not work right.

I have new installation of centos ready, so if you need any help I can 
do testing.

Regards,
Miha

On 8/9/2012 1:58 PM, Peter Olsson wrote:
> Steve, it's reported at: FS-4509
>
> The original change (commit 3818cae) mentioned that this problem was on CentOS. I haven't tried myself though.
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Steve Underwood
> Skickat: den 9 augusti 2012 13:42
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: Re: [Freeswitch-users] FS 1.2 make problems
>
> If 65ffaa8 causes the reported failure to compile, someone is using a
> broken compiler. Which platform is this?
>
> Steve
>
> On 08/09/2012 04:55 PM, Peter Olsson wrote:
>> Add to Jira please...
>>
>> FYI: It was fixed in commit  3818cae, but then reverted again in commit  65ffaa8.
>>
>> Submit the issue to Jira, and set Steve Underwood as the assignee, he's the author of libspandsp.
>>
>> /Peter
>>
>>
>> -----Ursprungligt meddelande-----
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
>> Skickat: den 9 augusti 2012 10:39
>> Till: FreeSWITCH Users Help
>> ?mne: [Freeswitch-users] FS 1.2 make problems
>>
>> HI,
>>
>> today I tried to install FS 1.2 on test server but after make I get this:
>>
>> make  all-am
>> make[3]: Entering directory `/usr/local/src/freeswitch/libs/curl/src'
>> make[3]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
>> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
>> make[2]: Entering directory `/usr/local/src/freeswitch/libs/curl'
>> make[2]: Nothing to be done for `all-am'.
>> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
>> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
>> cat /usr/local/src/freeswitch/src/include/switch_cpp.h | perl /usr/local/src/freeswitch/build/strip.pl > /usr/local/src/freeswitch/src/include/switch_swigable_cpp.h
>> make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
>> cc1: warnings being treated as errors
>> In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:114,
>>                     from ./src/include/private/switch_core_pvt.h:35,
>>                     from src/switch.c:48:
>> /usr/local/src/freeswitch/libs/spandsp/src/spandsp/t30.h:381: warning: ISO C restricts enumerator values to range of ???int???
>> make[1]: *** [freeswitch-switch.o] Error 1
>> make: *** [all] Error 2
>> You have mail in /var/spool/mail/root
>>
>> Thanks!
>>
>> Miha
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>  
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:5023a00532762097718087!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From joe.jflemmings at gmail.com  Thu Aug  9 16:44:31 2012
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Thu, 9 Aug 2012 05:44:31 -0700
Subject: [Freeswitch-users] CODEC event in Mod event socket
In-Reply-To: <CAPTXd0o8sRHzRQ7noQTn+5p54vf4ttzTyrFk0BDq5FNLZwMRsA@mail.gmail.com>
References: <CALH5ZrY5__hkX+rtoVfCa46r+q5vEhkwqKVZZTAWZ0Z4TmmEjA@mail.gmail.com>
	<CAPTXd0o8sRHzRQ7noQTn+5p54vf4ttzTyrFk0BDq5FNLZwMRsA@mail.gmail.com>
Message-ID: <CALH5ZrZ5Bq3g_6b-axxpgXSBvRFOc_0wt7pq08NC73xnR_oWow@mail.gmail.com>

I'm originating the call from the CLI using originate
http://wiki.freeswitch.org/wiki/Mod_spandsp#For_transmitting_a_fax_2
I can bet you that this is the first time i see this uniqueu id in the
events output.



On Thu, Aug 9, 2012 at 4:27 AM, Godson Gera <godson.g at gmail.com> wrote:

> You must already be having a channel in the FS with that unique id. Are
> you originating these calls from event socket ?
>
> On Thu, Aug 9, 2012 at 9:40 AM, Joe Flemmings <joe.jflemmings at gmail.com>wrote:
>
>> I?m using Inbound ?Mod event socket? and want to associate the codec
>> event with a call in the system.
>>
>> When the event is first fired by FreeSwitch, it has a new Unique-ID and
>> there is no way to associate it with any other channels.
>>
>> How can i associate this event so i know what call it belongs to. Please
>> see sample event bellow on issue. Unique-ID is a new value of
>> cdee98ff-d786-46be-b058-a29e5eddb0af. I would like to associate this event
>> with a channel in the system or call leg.
>>
>>
>> Event-Name: CODEC
>> Core-UUID: 45e18440-cf99-49ab-bc6e-d85c083312c5
>> FreeSWITCH-Hostname: freeswitch-64bit-lab
>> FreeSWITCH-Switchname: freeswitch-64bit-lab
>> FreeSWITCH-IPv4: 10.10.78.36
>> FreeSWITCH-IPv6: %3A%3A1
>> Event-Date-Local: 2012-08-08%2020%3A17%3A49
>> Event-Date-GMT: Thu,%2009%20Aug%202012%2003%3A17%3A49%20GMT
>> Event-Date-Timestamp: 1344482269907740
>> Event-Calling-File: switch_core_codec.c
>> Event-Calling-Function: switch_core_session_set_real_read_codec
>> Event-Calling-Line-Number: 162
>> Event-Sequence: 388579
>> Channel-State: CS_CONSUME_MEDIA
>> Channel-Call-State: RINGING
>> Channel-State-Number: 7
>> Channel-Name: sofia/external_10.10.78.36/fax_17144301800
>> _8518154256_8518154256_pbxcluster_4500847326%4010.10.78.36
>> Unique-ID: cdee98ff-d786-46be-b058-a29e5eddb0af
>> Call-Direction: outbound
>> Presence-Call-Direction: outbound
>> Channel-HIT-Dialplan: false
>> Channel-Call-UUID: cdee98ff-d786-46be-b058-a29e5eddb0af
>> Answer-State: ringing
>> Channel-Read-Codec-Name: PCMU
>> Channel-Read-Codec-Rate: 8000
>> Channel-Read-Codec-Bit-Rate: 64000
>> Channel-Write-Codec-Name: PCMU
>> Channel-Write-Codec-Rate: 8000
>> Channel-Write-Codec-Bit-Rate: 64000
>> Caller-Direction: outbound
>> Caller-Caller-ID-Name: 7144301800
>> Caller-Caller-ID-Number: 7144301800
>> Caller-Callee-ID-Name: Outbound%20Call
>> Caller-Callee-ID-Number: 17144301800
>> Caller-Network-Addr: 10.10.78.36
>> Caller-Destination-Number: fax_17144301800
>> _8518154256_8518154256_pbxcluster_4500847326
>> Caller-Unique-ID: cdee98ff-d786-46be-b058-a29e5eddb0af
>> Caller-Source: src/switch_ivr_originate.c
>> Caller-Context: default
>> Caller-Channel-Name: sofia/external_10.10.78.36/fax_17144301800
>> _8518154256_8518154256_pbxcluster_4500847326%4010.10.78.36
>> Caller-Profile-Index: 1
>> Caller-Profile-Created-Time: 1344482263028410
>> Caller-Channel-Created-Time: 1344482263028410
>> 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
>> channel-read-codec-name: PCMU
>> channel-read-codec-rate: 8000
>> channel-read-codec-bit-rate: 64000
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Thanks & Regards,
> Godson Gera
> FreeSWITCH Consultant <http://godson.in>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From godson.g at gmail.com  Thu Aug  9 17:28:51 2012
From: godson.g at gmail.com (Godson Gera)
Date: Thu, 9 Aug 2012 18:58:51 +0530
Subject: [Freeswitch-users] CODEC event in Mod event socket
In-Reply-To: <CALH5ZrZ5Bq3g_6b-axxpgXSBvRFOc_0wt7pq08NC73xnR_oWow@mail.gmail.com>
References: <CALH5ZrY5__hkX+rtoVfCa46r+q5vEhkwqKVZZTAWZ0Z4TmmEjA@mail.gmail.com>
	<CAPTXd0o8sRHzRQ7noQTn+5p54vf4ttzTyrFk0BDq5FNLZwMRsA@mail.gmail.com>
	<CALH5ZrZ5Bq3g_6b-axxpgXSBvRFOc_0wt7pq08NC73xnR_oWow@mail.gmail.com>
Message-ID: <CAPTXd0r568OO_oFBaLBKau972K5VoxaP2aKq53VUe0nfEPUcnw@mail.gmail.com>

Have you considered using origination_uuid channel variable where you can
specify your own UUID, that way you can recognize the first event related
to your originated call.

On Thu, Aug 9, 2012 at 6:14 PM, Joe Flemmings <joe.jflemmings at gmail.com>wrote:

> I'm originating the call from the CLI using originate
> http://wiki.freeswitch.org/wiki/Mod_spandsp#For_transmitting_a_fax_2
> I can bet you that this is the first time i see this uniqueu id in the
> events output.
>
>
>
>
> On Thu, Aug 9, 2012 at 4:27 AM, Godson Gera <godson.g at gmail.com> wrote:
>
>> You must already be having a channel in the FS with that unique id. Are
>> you originating these calls from event socket ?
>>
>> On Thu, Aug 9, 2012 at 9:40 AM, Joe Flemmings <joe.jflemmings at gmail.com>wrote:
>>
>>> I?m using Inbound ?Mod event socket? and want to associate the codec
>>> event with a call in the system.
>>>
>>> When the event is first fired by FreeSwitch, it has a new Unique-ID and
>>> there is no way to associate it with any other channels.
>>>
>>> How can i associate this event so i know what call it belongs to. Please
>>> see sample event bellow on issue. Unique-ID is a new value of
>>> cdee98ff-d786-46be-b058-a29e5eddb0af. I would like to associate this event
>>> with a channel in the system or call leg.
>>>
>>>
>>> Event-Name: CODEC
>>> Core-UUID: 45e18440-cf99-49ab-bc6e-d85c083312c5
>>> FreeSWITCH-Hostname: freeswitch-64bit-lab
>>> FreeSWITCH-Switchname: freeswitch-64bit-lab
>>> FreeSWITCH-IPv4: 10.10.78.36
>>> FreeSWITCH-IPv6: %3A%3A1
>>> Event-Date-Local: 2012-08-08%2020%3A17%3A49
>>> Event-Date-GMT: Thu,%2009%20Aug%202012%2003%3A17%3A49%20GMT
>>> Event-Date-Timestamp: 1344482269907740
>>> Event-Calling-File: switch_core_codec.c
>>> Event-Calling-Function: switch_core_session_set_real_read_codec
>>> Event-Calling-Line-Number: 162
>>> Event-Sequence: 388579
>>> Channel-State: CS_CONSUME_MEDIA
>>> Channel-Call-State: RINGING
>>> Channel-State-Number: 7
>>> Channel-Name: sofia/external_10.10.78.36/fax_17144301800
>>> _8518154256_8518154256_pbxcluster_4500847326%4010.10.78.36
>>> Unique-ID: cdee98ff-d786-46be-b058-a29e5eddb0af
>>> Call-Direction: outbound
>>> Presence-Call-Direction: outbound
>>> Channel-HIT-Dialplan: false
>>> Channel-Call-UUID: cdee98ff-d786-46be-b058-a29e5eddb0af
>>> Answer-State: ringing
>>> Channel-Read-Codec-Name: PCMU
>>> Channel-Read-Codec-Rate: 8000
>>> Channel-Read-Codec-Bit-Rate: 64000
>>> Channel-Write-Codec-Name: PCMU
>>> Channel-Write-Codec-Rate: 8000
>>> Channel-Write-Codec-Bit-Rate: 64000
>>> Caller-Direction: outbound
>>> Caller-Caller-ID-Name: 7144301800
>>> Caller-Caller-ID-Number: 7144301800
>>> Caller-Callee-ID-Name: Outbound%20Call
>>> Caller-Callee-ID-Number: 17144301800
>>> Caller-Network-Addr: 10.10.78.36
>>> Caller-Destination-Number: fax_17144301800
>>> _8518154256_8518154256_pbxcluster_4500847326
>>> Caller-Unique-ID: cdee98ff-d786-46be-b058-a29e5eddb0af
>>> Caller-Source: src/switch_ivr_originate.c
>>> Caller-Context: default
>>> Caller-Channel-Name: sofia/external_10.10.78.36/fax_17144301800
>>> _8518154256_8518154256_pbxcluster_4500847326%4010.10.78.36
>>> Caller-Profile-Index: 1
>>> Caller-Profile-Created-Time: 1344482263028410
>>> Caller-Channel-Created-Time: 1344482263028410
>>> 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
>>> channel-read-codec-name: PCMU
>>> channel-read-codec-rate: 8000
>>> channel-read-codec-bit-rate: 64000
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Thanks & Regards,
>> Godson Gera
>> FreeSWITCH Consultant <http://godson.in>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Thanks & Regards,
Godson Gera
FreeSWITCH Consultant <http://godson.in>
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From steveu at coppice.org  Thu Aug  9 18:03:12 2012
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 09 Aug 2012 22:03:12 +0800
Subject: [Freeswitch-users] FS 1.2 make problems
In-Reply-To: <5023A922.3070503@softnet.si>
References: <1FFF97C269757C458224B7C895F35F151414C0@cantor.std.visionutv.se>
	<5023A922.3070503@softnet.si>
Message-ID: <5023C320.1090304@coppice.org>

When this error was first reported in IRC it appeared to be because mu 
enum included a definition of 0x80000000 which is a negative number when 
considered as an integer. I thought that might be the issues, and 
changed to using 0x20000000. This was reported as fixing the issue. 
0x20000000 is most certainly a perfectly good positive integer value, 
and should be fine in an enum.

I just built the latest FreeSwitch on a 64 bit Centos 5.8 and a 64 bit 
Centos 6.3 machine. Both build OK. In fact, if I put the enum back to 
saying 0x80000000 it still builds OK. So, what is going on here?

Steve

On 08/09/2012 08:12 PM, Miha wrote:
> Steve, I tried to assigned ticket to you as Peter suggested but I guess
> it did not work right.
>
> I have new installation of centos ready, so if you need any help I can
> do testing.
>
> Regards,
> Miha
>
> On 8/9/2012 1:58 PM, Peter Olsson wrote:
>> Steve, it's reported at: FS-4509
>>
>> The original change (commit 3818cae) mentioned that this problem was on CentOS. I haven't tried myself though.
>>
>> /Peter
>>
>>
>> -----Ursprungligt meddelande-----
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Steve Underwood
>> Skickat: den 9 augusti 2012 13:42
>> Till: freeswitch-users at lists.freeswitch.org
>> ?mne: Re: [Freeswitch-users] FS 1.2 make problems
>>
>> If 65ffaa8 causes the reported failure to compile, someone is using a
>> broken compiler. Which platform is this?
>>
>> Steve
>>
>> On 08/09/2012 04:55 PM, Peter Olsson wrote:
>>> Add to Jira please...
>>>
>>> FYI: It was fixed in commit  3818cae, but then reverted again in commit  65ffaa8.
>>>
>>> Submit the issue to Jira, and set Steve Underwood as the assignee, he's the author of libspandsp.
>>>
>>> /Peter
>>>
>>>
>>> -----Ursprungligt meddelande-----
>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
>>> Skickat: den 9 augusti 2012 10:39
>>> Till: FreeSWITCH Users Help
>>> ?mne: [Freeswitch-users] FS 1.2 make problems
>>>
>>> HI,
>>>
>>> today I tried to install FS 1.2 on test server but after make I get this:
>>>
>>> make  all-am
>>> make[3]: Entering directory `/usr/local/src/freeswitch/libs/curl/src'
>>> make[3]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
>>> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl/src'
>>> make[2]: Entering directory `/usr/local/src/freeswitch/libs/curl'
>>> make[2]: Nothing to be done for `all-am'.
>>> make[2]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
>>> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/curl'
>>> cat /usr/local/src/freeswitch/src/include/switch_cpp.h | perl /usr/local/src/freeswitch/build/strip.pl > /usr/local/src/freeswitch/src/include/switch_swigable_cpp.h
>>> make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
>>> cc1: warnings being treated as errors
>>> In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:114,
>>>                      from ./src/include/private/switch_core_pvt.h:35,
>>>                      from src/switch.c:48:
>>> /usr/local/src/freeswitch/libs/spandsp/src/spandsp/t30.h:381: warning: ISO C restricts enumerator values to range of ???int???
>>> make[1]: *** [freeswitch-switch.o] Error 1
>>> make: *** [all] Error 2
>>> You have mail in /var/spool/mail/root
>>>
>>> Thanks!
>>>
>>> Miha
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>>  
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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:5023a00532762097718087!
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From godson.g at gmail.com  Thu Aug  9 19:14:32 2012
From: godson.g at gmail.com (Godson Gera)
Date: Thu, 9 Aug 2012 20:44:32 +0530
Subject: [Freeswitch-users] Calling all Python users @ ClueCon
In-Reply-To: <CAMwZ-tKg64MBE8YfWNhyHpoXFnh1pcMF07M0qY3CvFkETYaqYw@mail.gmail.com>
References: <CAMwZ-tKg64MBE8YfWNhyHpoXFnh1pcMF07M0qY3CvFkETYaqYw@mail.gmail.com>
Message-ID: <CAPTXd0rHiueX0_3-7yn4c5YdPGoE_uuFRdKYA2e4XpioK__tOg@mail.gmail.com>

Ufortunately, I am not at ClueCon , lets hangout here ;)

On Tue, Aug 7, 2012 at 10:02 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> Esteemed Pythonistas at ClueCon,
>
> Any Python users want to find an evening to talk shop and hang out at
> ClueCon this year? We planned on this last year, but we dropped the
> ball :) Anyway, let me know!
>
> Best,
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thanks & Regards,
Godson Gera
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From mitch.capper at gmail.com  Thu Aug  9 20:33:28 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 9 Aug 2012 09:33:28 -0700
Subject: [Freeswitch-users] TLS on FreeSwitch not Working
In-Reply-To: <BAY156-W11F916CCDCFC6B2A6B3E2DD4CC0@phx.gbl>
References: <BAY156-W11F916CCDCFC6B2A6B3E2DD4CC0@phx.gbl>
Message-ID: <CALn233rwaSGbtbcOV8sJPRA561T9MoWooBRYk4E=RNuvq9UM+Q@mail.gmail.com>

Turn on sofia tport logging it will tell you what its unable to setup
the TLS connection.

~Mitch

On Wed, Aug 8, 2012 at 10:09 PM, R W <wingcomm at hotmail.com> wrote:
> Hi All,
>
> I cannot seem to get TLS running on the sofia "internal" profile. Any
> assistance would be appreciated.
>
> I'm running FreeSWITCH Version 1.2.0-rc2+git~20120808T025758Z~9ac586adc8
> (1.2.0-rc2; git at commit 9ac586adc8 on Wed, 08 Aug 2012 02:57:58 Z) on
> Ubuntu 12.04 LTS.
>
> When I set internal_ssl_enable=true, and reload the sofia internal profile,
> I get the "usual" error:
>
> 2012-08-09 00:34:14.174431 [ERR] sofia.c:2289 Error Creating SIP UA for
> profile: internal
>
> <!-- SIP Profile: Internal -->
> <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=true"/>
> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>
>
> I verified that the OpenSSL development libraries were installed (Ubuntu
> package libssl-dev) and looked for references to ssl in the output from the
> compilation process and saw this:
>
> "checking for openssl... yes
>
> checking openssl_CFLAGS...
>
> checking openssl_LIBS... -lssl -lcrypto
>
>   adding "-DHAVE_OPENSSL" to SWITCH_AM_CFLAGS"
>
>
> ...
>
>
> "checking OpenSSL options with pkg-config... found
>
> checking for gdi32... no
>
> checking for CRYPTO_lock in -lcrypto... yes
>
> checking for SSL_connect in -lssl... yes
>
> checking openssl/x509.h usability... yes
>
> checking openssl/x509.h presence... yes
>
> checking for openssl/x509.h... yes
>
> checking openssl/rsa.h usability... yes
>
> checking openssl/rsa.h presence... yes
>
> checking for openssl/rsa.h... yes
>
> checking openssl/crypto.h usability... yes
>
> checking openssl/crypto.h presence... yes
>
> checking for openssl/crypto.h... yes
>
> checking openssl/pem.h usability... yes
>
> checking openssl/pem.h presence... yes
>
> checking for openssl/pem.h... yes
>
> checking openssl/ssl.h usability... yes
>
> checking openssl/ssl.h presence... yes
>
> checking for openssl/ssl.h... yes
>
> checking openssl/err.h usability... yes
>
> checking openssl/err.h presence... yes
>
> checking for openssl/err.h... yes
>
> checking openssl/pkcs12.h usability... yes
>
> checking openssl/pkcs12.h presence... yes
>
> checking for openssl/pkcs12.h... yes
>
> checking for ENGINE_init... yes
>
> checking openssl/engine.h usability... yes
>
> checking openssl/engine.h presence... yes
>
> checking for openssl/engine.h... yes
>
> checking for ENGINE_load_builtin_engines... yes
>
> checking for RAND_status... yes
>
> checking for RAND_screen... no
>
> checking for RAND_egd... yes
>
> checking for CRYPTO_cleanup_all_ex_data... yes
>
> checking for "/dev/urandom"... yes
>
> checking CA cert bundle install path...
> ${prefix}/share/curl/curl-ca-bundle.crt
>
> checking for inflateEnd in -lz... yes"
>
>
> Is there anything else I should be checking. Does freeswitch send logs
> anywhere other than ../freeswitch/log/ ?
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From joe.jflemmings at gmail.com  Thu Aug  9 21:28:14 2012
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Thu, 9 Aug 2012 10:28:14 -0700
Subject: [Freeswitch-users] CODEC event in Mod event socket
In-Reply-To: <CAPTXd0r568OO_oFBaLBKau972K5VoxaP2aKq53VUe0nfEPUcnw@mail.gmail.com>
References: <CALH5ZrY5__hkX+rtoVfCa46r+q5vEhkwqKVZZTAWZ0Z4TmmEjA@mail.gmail.com>
	<CAPTXd0o8sRHzRQ7noQTn+5p54vf4ttzTyrFk0BDq5FNLZwMRsA@mail.gmail.com>
	<CALH5ZrZ5Bq3g_6b-axxpgXSBvRFOc_0wt7pq08NC73xnR_oWow@mail.gmail.com>
	<CAPTXd0r568OO_oFBaLBKau972K5VoxaP2aKq53VUe0nfEPUcnw@mail.gmail.com>
Message-ID: <CALH5Zrbrz6F=n4VRh3zxU0zHGikVMQZYbmnhb7n5o8_g8gKKZg@mail.gmail.com>

Will try that out Godson. I could always tell the unique id of leg A and
leg B but when the code codec event is fired, it has a new unique ID and
their is no way to relate it to any of the call legs. Will try out
origination_uuid to see if any different.


Joe

On Thu, Aug 9, 2012 at 6:28 AM, Godson Gera <godson.g at gmail.com> wrote:

> Have you considered using origination_uuid channel variable where you can
> specify your own UUID, that way you can recognize the first event related
> to your originated call.
>
>
> On Thu, Aug 9, 2012 at 6:14 PM, Joe Flemmings <joe.jflemmings at gmail.com>wrote:
>
>> I'm originating the call from the CLI using originate
>> http://wiki.freeswitch.org/wiki/Mod_spandsp#For_transmitting_a_fax_2
>> I can bet you that this is the first time i see this uniqueu id in the
>> events output.
>>
>>
>>
>>
>> On Thu, Aug 9, 2012 at 4:27 AM, Godson Gera <godson.g at gmail.com> wrote:
>>
>>> You must already be having a channel in the FS with that unique id. Are
>>> you originating these calls from event socket ?
>>>
>>> On Thu, Aug 9, 2012 at 9:40 AM, Joe Flemmings <joe.jflemmings at gmail.com>wrote:
>>>
>>>> I?m using Inbound ?Mod event socket? and want to associate the codec
>>>> event with a call in the system.
>>>>
>>>> When the event is first fired by FreeSwitch, it has a new Unique-ID and
>>>> there is no way to associate it with any other channels.
>>>>
>>>> How can i associate this event so i know what call it belongs to.
>>>> Please see sample event bellow on issue. Unique-ID is a new value of
>>>> cdee98ff-d786-46be-b058-a29e5eddb0af. I would like to associate this event
>>>> with a channel in the system or call leg.
>>>>
>>>>
>>>> Event-Name: CODEC
>>>> Core-UUID: 45e18440-cf99-49ab-bc6e-d85c083312c5
>>>> FreeSWITCH-Hostname: freeswitch-64bit-lab
>>>> FreeSWITCH-Switchname: freeswitch-64bit-lab
>>>> FreeSWITCH-IPv4: 10.10.78.36
>>>> FreeSWITCH-IPv6: %3A%3A1
>>>> Event-Date-Local: 2012-08-08%2020%3A17%3A49
>>>> Event-Date-GMT: Thu,%2009%20Aug%202012%2003%3A17%3A49%20GMT
>>>> Event-Date-Timestamp: 1344482269907740
>>>> Event-Calling-File: switch_core_codec.c
>>>> Event-Calling-Function: switch_core_session_set_real_read_codec
>>>> Event-Calling-Line-Number: 162
>>>> Event-Sequence: 388579
>>>> Channel-State: CS_CONSUME_MEDIA
>>>> Channel-Call-State: RINGING
>>>> Channel-State-Number: 7
>>>> Channel-Name: sofia/external_10.10.78.36/fax_17144301800
>>>> _8518154256_8518154256_pbxcluster_4500847326%4010.10.78.36
>>>> Unique-ID: cdee98ff-d786-46be-b058-a29e5eddb0af
>>>> Call-Direction: outbound
>>>> Presence-Call-Direction: outbound
>>>> Channel-HIT-Dialplan: false
>>>> Channel-Call-UUID: cdee98ff-d786-46be-b058-a29e5eddb0af
>>>> Answer-State: ringing
>>>> Channel-Read-Codec-Name: PCMU
>>>> Channel-Read-Codec-Rate: 8000
>>>> Channel-Read-Codec-Bit-Rate: 64000
>>>> Channel-Write-Codec-Name: PCMU
>>>> Channel-Write-Codec-Rate: 8000
>>>> Channel-Write-Codec-Bit-Rate: 64000
>>>> Caller-Direction: outbound
>>>> Caller-Caller-ID-Name: 7144301800
>>>> Caller-Caller-ID-Number: 7144301800
>>>> Caller-Callee-ID-Name: Outbound%20Call
>>>> Caller-Callee-ID-Number: 17144301800
>>>> Caller-Network-Addr: 10.10.78.36
>>>> Caller-Destination-Number: fax_17144301800
>>>> _8518154256_8518154256_pbxcluster_4500847326
>>>> Caller-Unique-ID: cdee98ff-d786-46be-b058-a29e5eddb0af
>>>> Caller-Source: src/switch_ivr_originate.c
>>>> Caller-Context: default
>>>> Caller-Channel-Name: sofia/external_10.10.78.36/fax_17144301800
>>>> _8518154256_8518154256_pbxcluster_4500847326%4010.10.78.36
>>>> Caller-Profile-Index: 1
>>>> Caller-Profile-Created-Time: 1344482263028410
>>>> Caller-Channel-Created-Time: 1344482263028410
>>>> 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
>>>> channel-read-codec-name: PCMU
>>>> channel-read-codec-rate: 8000
>>>> channel-read-codec-bit-rate: 64000
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Thanks & Regards,
>>> Godson Gera
>>> FreeSWITCH Consultant <http://godson.in>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Thanks & Regards,
> Godson Gera
> FreeSWITCH Consultant <http://godson.in>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gabe at gundy.org  Thu Aug  9 22:42:23 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 9 Aug 2012 13:42:23 -0500
Subject: [Freeswitch-users] Calling all Python users @ ClueCon
In-Reply-To: <CAPTXd0rHiueX0_3-7yn4c5YdPGoE_uuFRdKYA2e4XpioK__tOg@mail.gmail.com>
References: <CAMwZ-tKg64MBE8YfWNhyHpoXFnh1pcMF07M0qY3CvFkETYaqYw@mail.gmail.com>
	<CAPTXd0rHiueX0_3-7yn4c5YdPGoE_uuFRdKYA2e4XpioK__tOg@mail.gmail.com>
Message-ID: <CAMwZ-tL6o4FMOUV-ePyY2a1=i-af3jT5LX63aqGL3iqRq5XQKw@mail.gmail.com>

On Thu, Aug 9, 2012 at 10:14 AM, Godson Gera <godson.g at gmail.com> wrote:
> Ufortunately, I am not at ClueCon , lets hangout here ;)

Nothing formal came together, but I did get a chance to chat with many
of the Python users, one on one. It's always good to get together and
talk shop :)

Gabe


From fiorix at gmail.com  Thu Aug  9 22:53:33 2012
From: fiorix at gmail.com (Alexandre Fiori)
Date: Thu, 9 Aug 2012 14:53:33 -0400
Subject: [Freeswitch-users] Calling all Python users @ ClueCon
In-Reply-To: <CAMwZ-tL6o4FMOUV-ePyY2a1=i-af3jT5LX63aqGL3iqRq5XQKw@mail.gmail.com>
References: <CAMwZ-tKg64MBE8YfWNhyHpoXFnh1pcMF07M0qY3CvFkETYaqYw@mail.gmail.com>
	<CAPTXd0rHiueX0_3-7yn4c5YdPGoE_uuFRdKYA2e4XpioK__tOg@mail.gmail.com>
	<CAMwZ-tL6o4FMOUV-ePyY2a1=i-af3jT5LX63aqGL3iqRq5XQKw@mail.gmail.com>
Message-ID: <4D4724E1-FD2F-440D-A196-47E209A10019@gmail.com>


i wish i was there :)


On 2012-08-09, at 2:42 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Thu, Aug 9, 2012 at 10:14 AM, Godson Gera <godson.g at gmail.com> wrote:
>> Ufortunately, I am not at ClueCon , lets hangout here ;)
> 
> Nothing formal came together, but I did get a chance to chat with many
> of the Python users, one on one. It's always good to get together and
> talk shop :)
> 
> Gabe
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


- af

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From moises.silva at gmail.com  Thu Aug  9 23:04:10 2012
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 9 Aug 2012 14:04:10 -0500
Subject: [Freeswitch-users] Calling all Python users @ ClueCon
In-Reply-To: <4D4724E1-FD2F-440D-A196-47E209A10019@gmail.com>
References: <CAMwZ-tKg64MBE8YfWNhyHpoXFnh1pcMF07M0qY3CvFkETYaqYw@mail.gmail.com>
	<CAPTXd0rHiueX0_3-7yn4c5YdPGoE_uuFRdKYA2e4XpioK__tOg@mail.gmail.com>
	<CAMwZ-tL6o4FMOUV-ePyY2a1=i-af3jT5LX63aqGL3iqRq5XQKw@mail.gmail.com>
	<4D4724E1-FD2F-440D-A196-47E209A10019@gmail.com>
Message-ID: <CAA4nhyA04Tm711mY_Lhd3smF3xaK4Q08snQVwCoQa29S_HqG1g@mail.gmail.com>

On Thu, Aug 9, 2012 at 1:53 PM, Alexandre Fiori <fiorix at gmail.com> wrote:

>
> i wish i was there :)
>

Hey dude, you should come next year, it is just a 1 hour flight from
Toronto :)

Moy
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From fiorix at gmail.com  Thu Aug  9 23:08:19 2012
From: fiorix at gmail.com (Alexandre Fiori)
Date: Thu, 9 Aug 2012 15:08:19 -0400
Subject: [Freeswitch-users] Calling all Python users @ ClueCon
In-Reply-To: <CAA4nhyA04Tm711mY_Lhd3smF3xaK4Q08snQVwCoQa29S_HqG1g@mail.gmail.com>
References: <CAMwZ-tKg64MBE8YfWNhyHpoXFnh1pcMF07M0qY3CvFkETYaqYw@mail.gmail.com>
	<CAPTXd0rHiueX0_3-7yn4c5YdPGoE_uuFRdKYA2e4XpioK__tOg@mail.gmail.com>
	<CAMwZ-tL6o4FMOUV-ePyY2a1=i-af3jT5LX63aqGL3iqRq5XQKw@mail.gmail.com>
	<4D4724E1-FD2F-440D-A196-47E209A10019@gmail.com>
	<CAA4nhyA04Tm711mY_Lhd3smF3xaK4Q08snQVwCoQa29S_HqG1g@mail.gmail.com>
Message-ID: <1C185E24-6545-410E-894B-258062F82EE5@gmail.com>


i'll do my best.. i wanted so badly to attend this year, but i'm busy with other stuff in philly


On 2012-08-09, at 3:04 PM, Moises Silva <moises.silva at gmail.com> wrote:

> On Thu, Aug 9, 2012 at 1:53 PM, Alexandre Fiori <fiorix at gmail.com> wrote:
> 
> i wish i was there :)
> 
> Hey dude, you should come next year, it is just a 1 hour flight from Toronto :)
> 
> Moy
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


- af

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From gabe at gundy.org  Thu Aug  9 23:30:14 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 9 Aug 2012 14:30:14 -0500
Subject: [Freeswitch-users] Calling all Python users @ ClueCon
In-Reply-To: <4D4724E1-FD2F-440D-A196-47E209A10019@gmail.com>
References: <CAMwZ-tKg64MBE8YfWNhyHpoXFnh1pcMF07M0qY3CvFkETYaqYw@mail.gmail.com>
	<CAPTXd0rHiueX0_3-7yn4c5YdPGoE_uuFRdKYA2e4XpioK__tOg@mail.gmail.com>
	<CAMwZ-tL6o4FMOUV-ePyY2a1=i-af3jT5LX63aqGL3iqRq5XQKw@mail.gmail.com>
	<4D4724E1-FD2F-440D-A196-47E209A10019@gmail.com>
Message-ID: <CAMwZ-tKLPu_op_eY9xHG0VVJNTS8vVdxGOeEvr0rJWd0_kekDA@mail.gmail.com>

On Thu, Aug 9, 2012 at 1:53 PM, Alexandre Fiori <fiorix at gmail.com> wrote:
> i wish i was there :)

Perhaps another year!

Gabe


From ajohnston at blimessaging.com  Fri Aug 10 00:55:18 2012
From: ajohnston at blimessaging.com (Adam Johnston)
Date: Thu, 9 Aug 2012 16:55:18 -0400
Subject: [Freeswitch-users] high load with ~200 fax calls despite low CPU
	utilization and low IO wait
Message-ID: <CAE2A9TCP0waoCDyB6aEkZ=viD5yUzJmnJpzsgFTA1UXAC6431w@mail.gmail.com>

Hi all,

My FreeSWITCH setup works as follow: I have a service that launches and
monitors faxes (via event sockets) on one of a few FreeSWITCH instances.
These FreeSWITCH instances are running on dual-core, 2ghz CentOS 5.8 and
CentOS 6.0 VMs. The issue is that once I get to ~200 simultaneous faxes on
any one FreeSWITCH VM the load average shoots up (typically ~30, although
I've seen it get much higher) and the VM becomes sluggish and occasionally
unusable.

top and sar output confirms that CPU utilization is pretty low, as are IO
wait and RAM usage.

I'm using a git head from 2012-02-06, although I see the same problem if
update to a more current head. On the most recent head I tried (2012-08-06)
the issue is worse, and crashes with a backtrace similar to this Jira,
http://jira.freeswitch.org/browse/FS-2893 despite my compiling Lua and
including the FreeSWITCH bindings.

These issues occur on both the CentOS 5.8 and 6.0 VMs. I'm in the process
of loading 5.8 onto a physical machine and testing there. It's also worth
nothing that this load issue does not occur when I launch faxes from a
script on one of the VMs.

Has anyone else seen similar high load issues before?

Many thanks,
Adam Johnston
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From asaad2 at gmail.com  Fri Aug 10 01:33:51 2012
From: asaad2 at gmail.com (BookBag)
Date: Thu, 9 Aug 2012 16:33:51 -0500
Subject: [Freeswitch-users] high load with ~200 fax calls despite low
 CPU utilization and low IO wait
In-Reply-To: <CAE2A9TCP0waoCDyB6aEkZ=viD5yUzJmnJpzsgFTA1UXAC6431w@mail.gmail.com>
References: <CAE2A9TCP0waoCDyB6aEkZ=viD5yUzJmnJpzsgFTA1UXAC6431w@mail.gmail.com>
Message-ID: <CAAdTMa-Ly6MOprXpqMQeEGyJx863CTYqL8tx4a3So+1hw-fXkQ@mail.gmail.com>

I'm running freeswitch also on a vm running centos 6.3 with fusionpbx. I
can vouch for the sluggishness. What I did is expand the memory. Try
expanding the memory to 4gb it should help but unfortunately if your on vm
your always going to get sub par performance.
On Aug 9, 2012 4:58 PM, "Adam Johnston" <ajohnston at blimessaging.com> wrote:

> Hi all,
>
> My FreeSWITCH setup works as follow: I have a service that launches and
> monitors faxes (via event sockets) on one of a few FreeSWITCH instances.
> These FreeSWITCH instances are running on dual-core, 2ghz CentOS 5.8 and
> CentOS 6.0 VMs. The issue is that once I get to ~200 simultaneous faxes on
> any one FreeSWITCH VM the load average shoots up (typically ~30, although
> I've seen it get much higher) and the VM becomes sluggish and occasionally
> unusable.
>
> top and sar output confirms that CPU utilization is pretty low, as are IO
> wait and RAM usage.
>
> I'm using a git head from 2012-02-06, although I see the same problem if
> update to a more current head. On the most recent head I tried (2012-08-06)
> the issue is worse, and crashes with a backtrace similar to this Jira,
> http://jira.freeswitch.org/browse/FS-2893 despite my compiling Lua and
> including the FreeSWITCH bindings.
>
> These issues occur on both the CentOS 5.8 and 6.0 VMs. I'm in the process
> of loading 5.8 onto a physical machine and testing there. It's also worth
> nothing that this load issue does not occur when I launch faxes from a
> script on one of the VMs.
>
> Has anyone else seen similar high load issues before?
>
> Many thanks,
> Adam Johnston
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Aug 10 02:57:17 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 9 Aug 2012 15:57:17 -0700
Subject: [Freeswitch-users] high load with ~200 fax calls despite low
 CPU utilization and low IO wait
In-Reply-To: <CAAdTMa-Ly6MOprXpqMQeEGyJx863CTYqL8tx4a3So+1hw-fXkQ@mail.gmail.com>
References: <CAE2A9TCP0waoCDyB6aEkZ=viD5yUzJmnJpzsgFTA1UXAC6431w@mail.gmail.com>
	<CAAdTMa-Ly6MOprXpqMQeEGyJx863CTYqL8tx4a3So+1hw-fXkQ@mail.gmail.com>
Message-ID: <CALn233pdj_+Q5RyVkad2Gmw=wV-mEMr-gWjN98f=zJSYByfujA@mail.gmail.com>

You should look to your VM solution provider for assistance(most have
their own performance tools) traditional linux tools do not always
provide proper load information inside of a VM.   Also look at the
load details on the host server to see where its straining generally
when your VM is un responsive its due to he host.

~Mitch

On Thu, Aug 9, 2012 at 2:33 PM, BookBag <asaad2 at gmail.com> wrote:
> I'm running freeswitch also on a vm running centos 6.3 with fusionpbx. I can
> vouch for the sluggishness. What I did is expand the memory. Try expanding
> the memory to 4gb it should help but unfortunately if your on vm your always
> going to get sub par performance.
>
> On Aug 9, 2012 4:58 PM, "Adam Johnston" <ajohnston at blimessaging.com> wrote:
>>
>> Hi all,
>>
>> My FreeSWITCH setup works as follow: I have a service that launches and
>> monitors faxes (via event sockets) on one of a few FreeSWITCH instances.
>> These FreeSWITCH instances are running on dual-core, 2ghz CentOS 5.8 and
>> CentOS 6.0 VMs. The issue is that once I get to ~200 simultaneous faxes on
>> any one FreeSWITCH VM the load average shoots up (typically ~30, although
>> I've seen it get much higher) and the VM becomes sluggish and occasionally
>> unusable.
>>
>> top and sar output confirms that CPU utilization is pretty low, as are IO
>> wait and RAM usage.
>>
>> I'm using a git head from 2012-02-06, although I see the same problem if
>> update to a more current head. On the most recent head I tried (2012-08-06)
>> the issue is worse, and crashes with a backtrace similar to this Jira,
>> http://jira.freeswitch.org/browse/FS-2893 despite my compiling Lua and
>> including the FreeSWITCH bindings.
>>
>> These issues occur on both the CentOS 5.8 and 6.0 VMs. I'm in the process
>> of loading 5.8 onto a physical machine and testing there. It's also worth
>> nothing that this load issue does not occur when I launch faxes from a
>> script on one of the VMs.
>>
>> Has anyone else seen similar high load issues before?
>>
>> Many thanks,
>> Adam Johnston
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Fri Aug 10 03:44:17 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Fri, 10 Aug 2012 01:44:17 +0200 (CEST)
Subject: [Freeswitch-users] Lua problem digits limit error
In-Reply-To: <CALn233pdj_+Q5RyVkad2Gmw=wV-mEMr-gWjN98f=zJSYByfujA@mail.gmail.com>
References: <CAE2A9TCP0waoCDyB6aEkZ=viD5yUzJmnJpzsgFTA1UXAC6431w@mail.gmail.com>
	<CAAdTMa-Ly6MOprXpqMQeEGyJx863CTYqL8tx4a3So+1hw-fXkQ@mail.gmail.com>
	<CALn233pdj_+Q5RyVkad2Gmw=wV-mEMr-gWjN98f=zJSYByfujA@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1208100139560.28153@1a-8140.antagus.de>

Hi guys, i like to define a counter
my script:
nummer_pointer  = argv[1];
api         = freeswitch.API();
msg_pointerdb = api:execute("hash", "select/mailbox_pointer_msg/msg");
  if msg_pointerdb == "nil" then
msg_pointerdb_add = api:execute("hash", 
"insert/mailbox_pointer_msg/msg/1");
elseif msg_pointerdb > 200  then
print(msg_pointerdb)
   else
msg_pointerdb = api:execute("hash", "select/mailbox_pointer_msg/msg"  )
pointer  =  msg_pointerdb  + nummer_pointer
  msg_pointerdb_add = api:execute("hash", "insert/mailbox_pointer_msg/msg/"
  ..pointer )
end

I become the error:
2012-08-10 01:41:35.499182 [ERR] mod_lua.cpp:198 
/usr/local/freeswitch/scripts/t
est-elseif.lua:6: attempt to compare number with string

what is wrong please?
thanks.


m

---------------
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 jmesquita at freeswitch.org  Fri Aug 10 06:07:05 2012
From: jmesquita at freeswitch.org (Jmesquita@freeswitch.org)
Date: Thu, 9 Aug 2012 23:07:05 -0300
Subject: [Freeswitch-users] Calling all Python users @ ClueCon
In-Reply-To: <CAMwZ-tKLPu_op_eY9xHG0VVJNTS8vVdxGOeEvr0rJWd0_kekDA@mail.gmail.com>
References: <CAMwZ-tKg64MBE8YfWNhyHpoXFnh1pcMF07M0qY3CvFkETYaqYw@mail.gmail.com>
	<CAPTXd0rHiueX0_3-7yn4c5YdPGoE_uuFRdKYA2e4XpioK__tOg@mail.gmail.com>
	<CAMwZ-tL6o4FMOUV-ePyY2a1=i-af3jT5LX63aqGL3iqRq5XQKw@mail.gmail.com>
	<4D4724E1-FD2F-440D-A196-47E209A10019@gmail.com>
	<CAMwZ-tKLPu_op_eY9xHG0VVJNTS8vVdxGOeEvr0rJWd0_kekDA@mail.gmail.com>
Message-ID: <464ABE0E-D1A9-4A3A-A36A-2EFDE0A77D43@freeswitch.org>

Make room for one more. It is for sure that you will have yet another pythonic freeswitch user on board.

Jo?o Mesquita

On 09/08/2012, at 04:30 p.m., Gabriel Gunderson <gabe at gundy.org> wrote:

> On Thu, Aug 9, 2012 at 1:53 PM, Alexandre Fiori <fiorix at gmail.com> wrote:
>> i wish i was there :)
> 
> Perhaps another year!
> 
> Gabe
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From govoiper at gmail.com  Fri Aug 10 08:55:50 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 10 Aug 2012 09:55:50 +0500
Subject: [Freeswitch-users] DTMF detection giving problems
Message-ID: <CAJUJwtiYJC_OdxahFZSZRFdbHcHxw37FQqJaVgTZsVW5kiGYDA@mail.gmail.com>

Hello All,

I've a small lua script which is just an IVR. Calls are landing on Asterisk
and then they are relayed over to FS. The problem is that the IVR don't
always detects what user pressed. What I've observed is that the DTMF
threshold is very high and some people just press slowly/quickly and the
tone duration gets low and FS don't detect anything.

I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've
also liberal-dtmf set in my sofia external.xml profile file which somewhat
got me better results but still DTMFs miss sometimes.

Please note that the asterisk on which the calls are landing have never
faced this issue before. Asterisk has PRI from which it receives the calls
and then relay to desired FS server.


Regards.
Sammy
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From govoiper at gmail.com  Fri Aug 10 09:30:57 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 10 Aug 2012 10:30:57 +0500
Subject: [Freeswitch-users] DTMF detection giving problems
In-Reply-To: <CAJUJwtiYJC_OdxahFZSZRFdbHcHxw37FQqJaVgTZsVW5kiGYDA@mail.gmail.com>
References: <CAJUJwtiYJC_OdxahFZSZRFdbHcHxw37FQqJaVgTZsVW5kiGYDA@mail.gmail.com>
Message-ID: <CAJUJwthrhTZUpLceykEaBDh8ue86Sq+6=9ytL3v=PQVG-4-RMQ@mail.gmail.com>

Narrowing down the problem, I see the PlayandGetDTMF starting the intended
audio file. Then user presses any DTMF and the PAGD breaks saying the user
pressed " " , yes and empty space :S.The variable which is supposed to hold
the DTMF has some space in it which it proudly logs to the console w/o
giving any errors !!




On Fri, Aug 10, 2012 at 9:55 AM, SamyGo <govoiper at gmail.com> wrote:

> Hello All,
>
> I've a small lua script which is just an IVR. Calls are landing on
> Asterisk and then they are relayed over to FS. The problem is that the IVR
> don't always detects what user pressed. What I've observed is that the DTMF
> threshold is very high and some people just press slowly/quickly and the
> tone duration gets low and FS don't detect anything.
>
> I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've
> also liberal-dtmf set in my sofia external.xml profile file which somewhat
> got me better results but still DTMFs miss sometimes.
>
> Please note that the asterisk on which the calls are landing have never
> faced this issue before. Asterisk has PRI from which it receives the calls
> and then relay to desired FS server.
>
>
> Regards.
> Sammy
>
>
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From runtwistedrat at gmail.com  Fri Aug 10 09:09:21 2012
From: runtwistedrat at gmail.com (ivan baydin)
Date: Fri, 10 Aug 2012 11:09:21 +0600
Subject: [Freeswitch-users] DTMF detection giving problems
In-Reply-To: <CAJUJwtiYJC_OdxahFZSZRFdbHcHxw37FQqJaVgTZsVW5kiGYDA@mail.gmail.com>
References: <CAJUJwtiYJC_OdxahFZSZRFdbHcHxw37FQqJaVgTZsVW5kiGYDA@mail.gmail.com>
Message-ID: <CAFOW8v1KT1XNUG4Ce2Em+4Msj7ri556GNy_FjxV_50u2RdF1Cw@mail.gmail.com>

Hello Sammy.
should check whether the operator is not blocking dtmf
I had a situation - in the same statement it worked, on the other - not
processed incoming dtmf

2012/8/10 SamyGo <govoiper at gmail.com>

> Hello All,
>
> I've a small lua script which is just an IVR. Calls are landing on
> Asterisk and then they are relayed over to FS. The problem is that the IVR
> don't always detects what user pressed. What I've observed is that the DTMF
> threshold is very high and some people just press slowly/quickly and the
> tone duration gets low and FS don't detect anything.
>
> I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've
> also liberal-dtmf set in my sofia external.xml profile file which somewhat
> got me better results but still DTMFs miss sometimes.
>
> Please note that the asterisk on which the calls are landing have never
> faced this issue before. Asterisk has PRI from which it receives the calls
> and then relay to desired FS server.
>
>
> Regards.
> Sammy
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From runtwistedrat at gmail.com  Fri Aug 10 09:43:35 2012
From: runtwistedrat at gmail.com (ivan baydin)
Date: Fri, 10 Aug 2012 11:43:35 +0600
Subject: [Freeswitch-users] DTMF detection giving problems
In-Reply-To: <CAJUJwthrhTZUpLceykEaBDh8ue86Sq+6=9ytL3v=PQVG-4-RMQ@mail.gmail.com>
References: <CAJUJwtiYJC_OdxahFZSZRFdbHcHxw37FQqJaVgTZsVW5kiGYDA@mail.gmail.com>
	<CAJUJwthrhTZUpLceykEaBDh8ue86Sq+6=9ytL3v=PQVG-4-RMQ@mail.gmail.com>
Message-ID: <CAFOW8v0zxjTZD=Yc3eHWwd1DQ0-k8xiApzeYBxQvMEJ5jpyRFw@mail.gmail.com>

You can see the script?

2012/8/10 SamyGo <govoiper at gmail.com>

> Narrowing down the problem, I see the PlayandGetDTMF starting the intended
> audio file. Then user presses any DTMF and the PAGD breaks saying the user
> pressed " " , yes and empty space :S.The variable which is supposed to hold
> the DTMF has some space in it which it proudly logs to the console w/o
> giving any errors !!
>
>
>
>
> On Fri, Aug 10, 2012 at 9:55 AM, SamyGo <govoiper at gmail.com> wrote:
>
>> Hello All,
>>
>> I've a small lua script which is just an IVR. Calls are landing on
>> Asterisk and then they are relayed over to FS. The problem is that the IVR
>> don't always detects what user pressed. What I've observed is that the DTMF
>> threshold is very high and some people just press slowly/quickly and the
>> tone duration gets low and FS don't detect anything.
>>
>> I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've
>> also liberal-dtmf set in my sofia external.xml profile file which somewhat
>> got me better results but still DTMFs miss sometimes.
>>
>> Please note that the asterisk on which the calls are landing have never
>> faced this issue before. Asterisk has PRI from which it receives the calls
>> and then relay to desired FS server.
>>
>>
>> Regards.
>> Sammy
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From govoiper at gmail.com  Fri Aug 10 12:10:41 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 10 Aug 2012 13:10:41 +0500
Subject: [Freeswitch-users] DTMF detection giving problems
In-Reply-To: <CAFOW8v1KT1XNUG4Ce2Em+4Msj7ri556GNy_FjxV_50u2RdF1Cw@mail.gmail.com>
References: <CAJUJwtiYJC_OdxahFZSZRFdbHcHxw37FQqJaVgTZsVW5kiGYDA@mail.gmail.com>
	<CAFOW8v1KT1XNUG4Ce2Em+4Msj7ri556GNy_FjxV_50u2RdF1Cw@mail.gmail.com>
Message-ID: <CAJUJwtjj8N=Li3iJ2t4-Hy=vJXVuVUuBYt9U84ojTL1E4vC8mw@mail.gmail.com>

No Ivan , I've no issue at all with DTMFs ever. Like I said there is an
Asterisk server which used torun the same IVRs and on same setup and never
got such a problem.
But not I'm relaying calls from Asterisk to FS and thus I've a SIP peer in
between. Calls are coming in OK, I intermittently face this issue where
user presses DTMFs and those digits are not recognized by
FS-PlayAndGetDigits()
I've  lines like this in my LUA

 choice = session:playAndGetDigits(1, 1,1,7, "#",sound_files[1], "",
"[0-9]")
 freeswitch.consoleLog("INFO","User Entered "..choice.." for this Survey\n")

*Logs for the calls without dtmf here:*

[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:1280

--------------------- Another Example---------------------

[DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 0:960
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm

--------------------- Another Example---------------------

[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 6:2560
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm



*Log for successful DTMF collected is as follows:*

[DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:8800
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[DEBUG] switch_ivr_play_say.c:2034 Test Regex [1][0-9]
[INFO] switch_cpp.cpp:1227 User Entered 1 for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
20ms
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm
-------


So I can see a pattern here. But I can't figure out this behavior. Is it
because I've 1 try only ! and after just once playing of file PAGD() breaks
no matter it gets DTMF or not !?

Regards,
Sammy


On Fri, Aug 10, 2012 at 10:09 AM, ivan baydin <runtwistedrat at gmail.com>wrote:

> Hello Sammy.
> should check whether the operator is not blocking dtmf
> I had a situation - in the same statement it worked, on the other - not
> processed incoming dtmf
>
> 2012/8/10 SamyGo <govoiper at gmail.com>
>
>>  Hello All,
>>
>> I've a small lua script which is just an IVR. Calls are landing on
>> Asterisk and then they are relayed over to FS. The problem is that the IVR
>> don't always detects what user pressed. What I've observed is that the DTMF
>> threshold is very high and some people just press slowly/quickly and the
>> tone duration gets low and FS don't detect anything.
>>
>> I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've
>> also liberal-dtmf set in my sofia external.xml profile file which somewhat
>> got me better results but still DTMFs miss sometimes.
>>
>> Please note that the asterisk on which the calls are landing have never
>> faced this issue before. Asterisk has PRI from which it receives the calls
>> and then relay to desired FS server.
>>
>>
>> Regards.
>> Sammy
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Aug 10 12:23:12 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 10 Aug 2012 08:23:12 +0000
Subject: [Freeswitch-users] DTMF detection giving problems
Message-ID: <1FFF97C269757C458224B7C895F35F15141E5C@cantor.std.visionutv.se>

7 ms seems like a very short timeout to have?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r SamyGo
Skickat: den 10 augusti 2012 10:11
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] DTMF detection giving problems

No Ivan , I've no issue at all with DTMFs ever. Like I said there is an Asterisk server which used torun the same IVRs and on same setup and never got such a problem.
But not I'm relaying calls from Asterisk to FS and thus I've a SIP peer in between. Calls are coming in OK, I intermittently face this issue where user presses DTMFs and those digits are not recognized by FS-PlayAndGetDigits()
I've  lines like this in my LUA

 choice = session:playAndGetDigits(1, 1,1,7, "#",sound_files[1], "", "[0-9]")
 freeswitch.consoleLog("INFO","User Entered "..choice.." for this Survey\n")

Logs for the calls without dtmf here:

[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:1280

--------------------- Another Example---------------------

[DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 0:960
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm

--------------------- Another Example---------------------

[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 6:2560
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm



Log for successful DTMF collected is as follows:

[DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:8800
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[DEBUG] switch_ivr_play_say.c:2034 Test Regex [1][0-9]
[INFO] switch_cpp.cpp:1227 User Entered 1 for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm
-------


So I can see a pattern here. But I can't figure out this behavior. Is it because I've 1 try only ! and after just once playing of file PAGD() breaks no matter it gets DTMF or not !?

Regards,
Sammy


On Fri, Aug 10, 2012 at 10:09 AM, ivan baydin <runtwistedrat at gmail.com<mailto:runtwistedrat at gmail.com>> wrote:
Hello Sammy.
should check whether the operator is not blocking dtmf
I had a situation - in the same statement it worked, on the other - not processed incoming dtmf

2012/8/10 SamyGo <govoiper at gmail.com<mailto:govoiper at gmail.com>>
Hello All,

I've a small lua script which is just an IVR. Calls are landing on Asterisk and then they are relayed over to FS. The problem is that the IVR don't always detects what user pressed. What I've observed is that the DTMF threshold is very high and some people just press slowly/quickly and the tone duration gets low and FS don't detect anything.

I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've also liberal-dtmf set in my sofia external.xml profile file which somewhat got me better results but still DTMFs miss sometimes.

Please note that the asterisk on which the calls are landing have never faced this issue before. Asterisk has PRI from which it receives the calls and then relay to desired FS server.


Regards.
Sammy


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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:5024c05932762100017234!
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From govoiper at gmail.com  Fri Aug 10 12:30:55 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 10 Aug 2012 13:30:55 +0500
Subject: [Freeswitch-users] DTMF detection giving problems
In-Reply-To: <1FFF97C269757C458224B7C895F35F15141E5C@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F15141E5C@cantor.std.visionutv.se>
Message-ID: <CAJUJwtigx7zYqD4JC1k6fvojSN+1e7SGJKWcBLqwe8hgGc-CBA@mail.gmail.com>

is that ms ? God bless the poor callers !! Goodness. wait I'll be back.

On Fri, Aug 10, 2012 at 1:23 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  7 ms seems like a very short timeout to have?****
>
> ** **
>
> /Peter****
>
> ** **
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *SamyGo
> *Skickat:* den 10 augusti 2012 10:11
> *Till:* FreeSWITCH Users Help
> *?mne:* Re: [Freeswitch-users] DTMF detection giving problems****
>
> ** **
>
> No Ivan , I've no issue at all with DTMFs ever. Like I said there is an
> Asterisk server which used torun the same IVRs and on same setup and never
> got such a problem.****
>
> But not I'm relaying calls from Asterisk to FS and thus I've a SIP peer in
> between. Calls are coming in OK, I intermittently face this issue where
> user presses DTMFs and those digits are not recognized by
> FS-PlayAndGetDigits() ****
>
> I've  lines like this in my LUA****
>
> ** **
>
>  choice = session:playAndGetDigits(1, 1,1,7, "#",sound_files[1], "",
> "[0-9]")****
>
>  freeswitch.consoleLog("INFO","User Entered "..choice.." for this
> Survey\n")****
>
> ** **
>
> *Logs for the calls without dtmf here:*****
>
> ** **
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm***
> *
>
> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey****
>
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
> 20ms****
>
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:1280****
>
> ** **
>
> --------------------- Another Example---------------------****
>
> ** **
>
> [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm***
> *
>
> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey****
>
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
> 20ms****
>
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 0:960****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm***
> *
>
> ** **
>
> --------------------- Another Example---------------------****
>
> ** **
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm***
> *
>
> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey****
>
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
> 20ms****
>
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 6:2560****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm***
> *
>
> ** **
>
> ** **
>
> ** **
>
> *Log for successful DTMF collected is as follows:*****
>
> ** **
>
> [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.****
>
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:8800****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm***
> *
>
> [DEBUG] switch_ivr_play_say.c:2034 Test Regex [1][0-9]****
>
> [INFO] switch_cpp.cpp:1227 User Entered 1 for this Survey****
>
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
> 20ms****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm***
> *
>
> -------****
>
> ** **
>
> ** **
>
> So I can see a pattern here. But I can't figure out this behavior. Is it
> because I've 1 try only ! and after just once playing of file PAGD() breaks
> no matter it gets DTMF or not !?****
>
> ** **
>
> Regards,****
>
> Sammy****
>
> ** **
>
> ** **
>
> On Fri, Aug 10, 2012 at 10:09 AM, ivan baydin <runtwistedrat at gmail.com>
> wrote:****
>
> Hello Sammy.****
>
> should check whether the operator is not blocking dtmf****
>
> I had a situation - in the same statement it worked, on the other - not
> processed incoming dtmf ****
>
> ** **
>
> 2012/8/10 SamyGo <govoiper at gmail.com>****
>
> Hello All,****
>
> ** **
>
> I've a small lua script which is just an IVR. Calls are landing on
> Asterisk and then they are relayed over to FS. The problem is that the IVR
> don't always detects what user pressed. What I've observed is that the DTMF
> threshold is very high and some people just press slowly/quickly and the
> tone duration gets low and FS don't detect anything. ****
>
> ** **
>
> I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've
> also liberal-dtmf set in my sofia external.xml profile file which somewhat
> got me better results but still DTMFs miss sometimes.****
>
> ** **
>
> Please note that the asterisk on which the calls are landing have never
> faced this issue before. Asterisk has PRI from which it receives the calls
> and then relay to desired FS server.****
>
> ** **
>
> ** **
>
> Regards.****
>
> Sammy****
>
> ** **
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:5024c05932762100017234! ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From m.prevail at gmail.com  Fri Aug 10 12:36:09 2012
From: m.prevail at gmail.com (Prevail Magid)
Date: Fri, 10 Aug 2012 11:36:09 +0300
Subject: [Freeswitch-users] Send event when FS starting
Message-ID: <CANJXrA8ehM++E63rdJ7qYz96NeHu-MOf3VqDrJmvJi95kTkzog@mail.gmail.com>

Hello,

Is it possible to configure event_socket to send event like 'FREESWITCH has
been started' on outbound client?
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From govoiper at gmail.com  Fri Aug 10 12:38:13 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 10 Aug 2012 13:38:13 +0500
Subject: [Freeswitch-users] DTMF detection giving problems
In-Reply-To: <CAJUJwtigx7zYqD4JC1k6fvojSN+1e7SGJKWcBLqwe8hgGc-CBA@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F15141E5C@cantor.std.visionutv.se>
	<CAJUJwtigx7zYqD4JC1k6fvojSN+1e7SGJKWcBLqwe8hgGc-CBA@mail.gmail.com>
Message-ID: <CAJUJwti3pBAHni+Qrrm1pcdkH8VkEwiP+CRU0O=rRVekKju+UA@mail.gmail.com>

OK. Fixed that. Now, The thing is when I was testing it I never waited for
the menu to end and just pressed buttons normally. And even then I couldn't
get any DTMF.

On Fri, Aug 10, 2012 at 1:30 PM, SamyGo <govoiper at gmail.com> wrote:

> is that ms ? God bless the poor callers !! Goodness. wait I'll be back.
>
>
> On Fri, Aug 10, 2012 at 1:23 PM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
>>  7 ms seems like a very short timeout to have?****
>>
>> ** **
>>
>> /Peter****
>>
>> ** **
>>
>> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *F?r *SamyGo
>> *Skickat:* den 10 augusti 2012 10:11
>> *Till:* FreeSWITCH Users Help
>> *?mne:* Re: [Freeswitch-users] DTMF detection giving problems****
>>
>> ** **
>>
>> No Ivan , I've no issue at all with DTMFs ever. Like I said there is an
>> Asterisk server which used torun the same IVRs and on same setup and never
>> got such a problem.****
>>
>> But not I'm relaying calls from Asterisk to FS and thus I've a SIP peer
>> in between. Calls are coming in OK, I intermittently face this issue where
>> user presses DTMFs and those digits are not recognized by
>> FS-PlayAndGetDigits() ****
>>
>> I've  lines like this in my LUA****
>>
>> ** **
>>
>>  choice = session:playAndGetDigits(1, 1,1,7, "#",sound_files[1], "",
>> "[0-9]")****
>>
>>  freeswitch.consoleLog("INFO","User Entered "..choice.." for this
>> Survey\n")****
>>
>> ** **
>>
>> *Logs for the calls without dtmf here:*****
>>
>> ** **
>>
>> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm**
>> **
>>
>> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey****
>>
>> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
>> 20ms****
>>
>> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:1280****
>>
>> ** **
>>
>> --------------------- Another Example---------------------****
>>
>> ** **
>>
>> [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.****
>>
>> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm**
>> **
>>
>> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey****
>>
>> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
>> 20ms****
>>
>> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 0:960****
>>
>> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm**
>> **
>>
>> ** **
>>
>> --------------------- Another Example---------------------****
>>
>> ** **
>>
>> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm**
>> **
>>
>> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey****
>>
>> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
>> 20ms****
>>
>> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 6:2560****
>>
>> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm**
>> **
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> *Log for successful DTMF collected is as follows:*****
>>
>> ** **
>>
>> [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.****
>>
>> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:8800****
>>
>> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm**
>> **
>>
>> [DEBUG] switch_ivr_play_say.c:2034 Test Regex [1][0-9]****
>>
>> [INFO] switch_cpp.cpp:1227 User Entered 1 for this Survey****
>>
>> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
>> 20ms****
>>
>> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm**
>> **
>>
>> -------****
>>
>> ** **
>>
>> ** **
>>
>> So I can see a pattern here. But I can't figure out this behavior. Is it
>> because I've 1 try only ! and after just once playing of file PAGD() breaks
>> no matter it gets DTMF or not !?****
>>
>> ** **
>>
>> Regards,****
>>
>> Sammy****
>>
>> ** **
>>
>> ** **
>>
>> On Fri, Aug 10, 2012 at 10:09 AM, ivan baydin <runtwistedrat at gmail.com>
>> wrote:****
>>
>> Hello Sammy.****
>>
>> should check whether the operator is not blocking dtmf****
>>
>> I had a situation - in the same statement it worked, on the other - not
>> processed incoming dtmf ****
>>
>> ** **
>>
>> 2012/8/10 SamyGo <govoiper at gmail.com>****
>>
>> Hello All,****
>>
>> ** **
>>
>> I've a small lua script which is just an IVR. Calls are landing on
>> Asterisk and then they are relayed over to FS. The problem is that the IVR
>> don't always detects what user pressed. What I've observed is that the DTMF
>> threshold is very high and some people just press slowly/quickly and the
>> tone duration gets low and FS don't detect anything. ****
>>
>> ** **
>>
>> I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've
>> also liberal-dtmf set in my sofia external.xml profile file which somewhat
>> got me better results but still DTMFs miss sometimes.****
>>
>> ** **
>>
>> Please note that the asterisk on which the calls are landing have never
>> faced this issue before. Asterisk has PRI from which it receives the calls
>> and then relay to desired FS server.****
>>
>> ** **
>>
>> ** **
>>
>> Regards.****
>>
>> Sammy****
>>
>> ** **
>>
>> ** **
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>> ** **
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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:5024c05932762100017234! ****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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 Aug 10 12:44:31 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 10 Aug 2012 08:44:31 +0000
Subject: [Freeswitch-users] Send event when FS starting
Message-ID: <1FFF97C269757C458224B7C895F35F15141F21@cantor.std.visionutv.se>

The outbound socket is always related to a call, so that won't work.

If you use incoming ESL you'll see when it's up, since you're able to connect.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Prevail Magid
Skickat: den 10 augusti 2012 10:36
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Send event when FS starting

Hello,

Is it possible to configure event_socket to send event like 'FREESWITCH has been started' on outbound client? !DSPAM:5024c59432761952819871!
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From peter.olsson at visionutveckling.se  Fri Aug 10 12:56:32 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 10 Aug 2012 08:56:32 +0000
Subject: [Freeswitch-users] DTMF detection giving problems
Message-ID: <1FFF97C269757C458224B7C895F35F15141F91@cantor.std.visionutv.se>

Did you terminate with the "#" DTMF char?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r SamyGo
Skickat: den 10 augusti 2012 10:38
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] DTMF detection giving problems

OK. Fixed that. Now, The thing is when I was testing it I never waited for the menu to end and just pressed buttons normally. And even then I couldn't get any DTMF.

On Fri, Aug 10, 2012 at 1:30 PM, SamyGo <govoiper at gmail.com<mailto:govoiper at gmail.com>> wrote:
is that ms ? God bless the poor callers !! Goodness. wait I'll be back.

On Fri, Aug 10, 2012 at 1:23 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
7 ms seems like a very short timeout to have?

/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 SamyGo
Skickat: den 10 augusti 2012 10:11
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] DTMF detection giving problems

No Ivan , I've no issue at all with DTMFs ever. Like I said there is an Asterisk server which used torun the same IVRs and on same setup and never got such a problem.
But not I'm relaying calls from Asterisk to FS and thus I've a SIP peer in between. Calls are coming in OK, I intermittently face this issue where user presses DTMFs and those digits are not recognized by FS-PlayAndGetDigits()
I've  lines like this in my LUA

 choice = session:playAndGetDigits(1, 1,1,7, "#",sound_files[1], "", "[0-9]")
 freeswitch.consoleLog("INFO","User Entered "..choice.." for this Survey\n")

Logs for the calls without dtmf here:

[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:1280

--------------------- Another Example---------------------

[DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 0:960
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm

--------------------- Another Example---------------------

[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 6:2560
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm



Log for successful DTMF collected is as follows:

[DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:8800
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[DEBUG] switch_ivr_play_say.c:2034 Test Regex [1][0-9]
[INFO] switch_cpp.cpp:1227 User Entered 1 for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm
-------


So I can see a pattern here. But I can't figure out this behavior. Is it because I've 1 try only ! and after just once playing of file PAGD() breaks no matter it gets DTMF or not !?

Regards,
Sammy


On Fri, Aug 10, 2012 at 10:09 AM, ivan baydin <runtwistedrat at gmail.com<mailto:runtwistedrat at gmail.com>> wrote:
Hello Sammy.
should check whether the operator is not blocking dtmf
I had a situation - in the same statement it worked, on the other - not processed incoming dtmf

2012/8/10 SamyGo <govoiper at gmail.com<mailto:govoiper at gmail.com>>
Hello All,

I've a small lua script which is just an IVR. Calls are landing on Asterisk and then they are relayed over to FS. The problem is that the IVR don't always detects what user pressed. What I've observed is that the DTMF threshold is very high and some people just press slowly/quickly and the tone duration gets low and FS don't detect anything.

I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've also liberal-dtmf set in my sofia external.xml profile file which somewhat got me better results but still DTMFs miss sometimes.

Please note that the asterisk on which the calls are landing have never faced this issue before. Asterisk has PRI from which it receives the calls and then relay to desired FS server.


Regards.
Sammy


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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:5024c66532766219410110!
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From govoiper at gmail.com  Fri Aug 10 13:17:24 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 10 Aug 2012 14:17:24 +0500
Subject: [Freeswitch-users] DTMF detection giving problems
In-Reply-To: <1FFF97C269757C458224B7C895F35F15141F91@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F15141F91@cantor.std.visionutv.se>
Message-ID: <CAJUJwtgPM0+4piYH40tsNJi+WY1Xb=LZ40Mwrw5Dt9EAN3eoJQ@mail.gmail.com>

No, and basically that isn't needed - I only need to capture 1 digit only.
and as soon as one digit is captured it can go to next porgram-line.

BR
Sammy


On Fri, Aug 10, 2012 at 1:56 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  Did you terminate with the ?#? DTMF char?****
>
> ** **
>
> /Peter****
>
> ** **
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *SamyGo
> *Skickat:* den 10 augusti 2012 10:38
>
> *Till:* FreeSWITCH Users Help
> *?mne:* Re: [Freeswitch-users] DTMF detection giving problems****
>
> ** **
>
> OK. Fixed that. Now, The thing is when I was testing it I never waited for
> the menu to end and just pressed buttons normally. And even then I couldn't
> get any DTMF. ****
>
> ** **
>
> On Fri, Aug 10, 2012 at 1:30 PM, SamyGo <govoiper at gmail.com> wrote:****
>
> is that ms ? God bless the poor callers !! Goodness. wait I'll be back.***
> *
>
> ** **
>
> On Fri, Aug 10, 2012 at 1:23 PM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:****
>
> 7 ms seems like a very short timeout to have?****
>
>  ****
>
> /Peter****
>
>  ****
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *SamyGo
> *Skickat:* den 10 augusti 2012 10:11
> *Till:* FreeSWITCH Users Help
> *?mne:* Re: [Freeswitch-users] DTMF detection giving problems****
>
>  ****
>
> No Ivan , I've no issue at all with DTMFs ever. Like I said there is an
> Asterisk server which used torun the same IVRs and on same setup and never
> got such a problem.****
>
> But not I'm relaying calls from Asterisk to FS and thus I've a SIP peer in
> between. Calls are coming in OK, I intermittently face this issue where
> user presses DTMFs and those digits are not recognized by
> FS-PlayAndGetDigits() ****
>
> I've  lines like this in my LUA****
>
>  ****
>
>  choice = session:playAndGetDigits(1, 1,1,7, "#",sound_files[1], "",
> "[0-9]")****
>
>  freeswitch.consoleLog("INFO","User Entered "..choice.." for this
> Survey\n")****
>
>  ****
>
> *Logs for the calls without dtmf here:*****
>
>  ****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm***
> *
>
> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey****
>
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
> 20ms****
>
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:1280****
>
>  ****
>
> --------------------- Another Example---------------------****
>
>  ****
>
> [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm***
> *
>
> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey****
>
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
> 20ms****
>
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 0:960****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm***
> *
>
>  ****
>
> --------------------- Another Example---------------------****
>
>  ****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm***
> *
>
> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey****
>
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
> 20ms****
>
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 6:2560****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm***
> *
>
>  ****
>
>  ****
>
>  ****
>
> *Log for successful DTMF collected is as follows:*****
>
>  ****
>
> [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.****
>
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:8800****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm***
> *
>
> [DEBUG] switch_ivr_play_say.c:2034 Test Regex [1][0-9]****
>
> [INFO] switch_cpp.cpp:1227 User Entered 1 for this Survey****
>
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels
> 20ms****
>
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm***
> *
>
> -------****
>
>  ****
>
>  ****
>
> So I can see a pattern here. But I can't figure out this behavior. Is it
> because I've 1 try only ! and after just once playing of file PAGD() breaks
> no matter it gets DTMF or not !?****
>
>  ****
>
> Regards,****
>
> Sammy****
>
>  ****
>
>  ****
>
> On Fri, Aug 10, 2012 at 10:09 AM, ivan baydin <runtwistedrat at gmail.com>
> wrote:****
>
> Hello Sammy.****
>
> should check whether the operator is not blocking dtmf****
>
> I had a situation - in the same statement it worked, on the other - not
> processed incoming dtmf ****
>
>  ****
>
> 2012/8/10 SamyGo <govoiper at gmail.com>****
>
> Hello All,****
>
>  ****
>
> I've a small lua script which is just an IVR. Calls are landing on
> Asterisk and then they are relayed over to FS. The problem is that the IVR
> don't always detects what user pressed. What I've observed is that the DTMF
> threshold is very high and some people just press slowly/quickly and the
> tone duration gets low and FS don't detect anything. ****
>
>  ****
>
> I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've
> also liberal-dtmf set in my sofia external.xml profile file which somewhat
> got me better results but still DTMFs miss sometimes.****
>
>  ****
>
> Please note that the asterisk on which the calls are landing have never
> faced this issue before. Asterisk has PRI from which it receives the calls
> and then relay to desired FS server.****
>
>  ****
>
>  ****
>
> Regards.****
>
> Sammy****
>
>  ****
>
>  ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:5024c66532766219410110! ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Aug 10 13:31:00 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 10 Aug 2012 09:31:00 +0000
Subject: [Freeswitch-users] DTMF detection giving problems
Message-ID: <1FFF97C269757C458224B7C895F35F1514200C@cantor.std.visionutv.se>

Ok. I've never used this function, but it looks like the terminator you have given "#", should be replaced by "0123456789", to make it stop grabbing on first input digit. But as I said - I don't really know :)

Mvh
Peter Olsson

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r SamyGo
Skickat: den 10 augusti 2012 11:17
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] DTMF detection giving problems

No, and basically that isn't needed - I only need to capture 1 digit only. and as soon as one digit is captured it can go to next porgram-line.

BR
Sammy

On Fri, Aug 10, 2012 at 1:56 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Did you terminate with the "#" DTMF char?

/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 SamyGo
Skickat: den 10 augusti 2012 10:38

Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] DTMF detection giving problems

OK. Fixed that. Now, The thing is when I was testing it I never waited for the menu to end and just pressed buttons normally. And even then I couldn't get any DTMF.

On Fri, Aug 10, 2012 at 1:30 PM, SamyGo <govoiper at gmail.com<mailto:govoiper at gmail.com>> wrote:
is that ms ? God bless the poor callers !! Goodness. wait I'll be back.

On Fri, Aug 10, 2012 at 1:23 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
7 ms seems like a very short timeout to have?

/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 SamyGo
Skickat: den 10 augusti 2012 10:11
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] DTMF detection giving problems

No Ivan , I've no issue at all with DTMFs ever. Like I said there is an Asterisk server which used torun the same IVRs and on same setup and never got such a problem.
But not I'm relaying calls from Asterisk to FS and thus I've a SIP peer in between. Calls are coming in OK, I intermittently face this issue where user presses DTMFs and those digits are not recognized by FS-PlayAndGetDigits()
I've  lines like this in my LUA

 choice = session:playAndGetDigits(1, 1,1,7, "#",sound_files[1], "", "[0-9]")
 freeswitch.consoleLog("INFO","User Entered "..choice.." for this Survey\n")

Logs for the calls without dtmf here:

[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:1280

--------------------- Another Example---------------------

[DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 0:960
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm

--------------------- Another Example---------------------

[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[INFO] switch_cpp.cpp:1227 User Entered  for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 6:2560
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm



Log for successful DTMF collected is as follows:

[DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.
[DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:8800
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
[DEBUG] switch_ivr_play_say.c:2034 Test Regex [1][0-9]
[INFO] switch_cpp.cpp:1227 User Entered 1 for this Survey
[DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
[DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm
-------


So I can see a pattern here. But I can't figure out this behavior. Is it because I've 1 try only ! and after just once playing of file PAGD() breaks no matter it gets DTMF or not !?

Regards,
Sammy


On Fri, Aug 10, 2012 at 10:09 AM, ivan baydin <runtwistedrat at gmail.com<mailto:runtwistedrat at gmail.com>> wrote:
Hello Sammy.
should check whether the operator is not blocking dtmf
I had a situation - in the same statement it worked, on the other - not processed incoming dtmf

2012/8/10 SamyGo <govoiper at gmail.com<mailto:govoiper at gmail.com>>
Hello All,

I've a small lua script which is just an IVR. Calls are landing on Asterisk and then they are relayed over to FS. The problem is that the IVR don't always detects what user pressed. What I've observed is that the DTMF threshold is very high and some people just press slowly/quickly and the tone duration gets low and FS don't detect anything.

I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've also liberal-dtmf set in my sofia external.xml profile file which somewhat got me better results but still DTMFs miss sometimes.

Please note that the asterisk on which the calls are landing have never faced this issue before. Asterisk has PRI from which it receives the calls and then relay to desired FS server.


Regards.
Sammy


_________________________________________________________________________
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http://www.freeswitchsolutions.com




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From covici at ccs.covici.com  Fri Aug 10 14:45:14 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 10 Aug 2012 06:45:14 -0400
Subject: [Freeswitch-users] DTMF detection giving problems
In-Reply-To: <1FFF97C269757C458224B7C895F35F1514200C@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1514200C@cantor.std.visionutv.se>
Message-ID: <10698.1344595514@ccs.covici.com>

Not necessary to change the termination -- I do this all the time, but I
have a analog phone directly tied to fs through an ata and it detects
the dtmf fine.

Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> Ok. I've never used this function, but it looks like the terminator you have given "#", should be replaced by "0123456789", to make it stop grabbing on first input digit. But as I said - I don't really know :)
> 
> Mvh
> Peter Olsson
> 
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r SamyGo
> Skickat: den 10 augusti 2012 11:17
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] DTMF detection giving problems
> 
> No, and basically that isn't needed - I only need to capture 1 digit only. and as soon as one digit is captured it can go to next porgram-line.
> 
> BR
> Sammy
> 
> On Fri, Aug 10, 2012 at 1:56 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
> Did you terminate with the "#" DTMF char?
> 
> /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 SamyGo
> Skickat: den 10 augusti 2012 10:38
> 
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] DTMF detection giving problems
> 
> OK. Fixed that. Now, The thing is when I was testing it I never waited for the menu to end and just pressed buttons normally. And even then I couldn't get any DTMF.
> 
> On Fri, Aug 10, 2012 at 1:30 PM, SamyGo <govoiper at gmail.com<mailto:govoiper at gmail.com>> wrote:
> is that ms ? God bless the poor callers !! Goodness. wait I'll be back.
> 
> On Fri, Aug 10, 2012 at 1:23 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
> 7 ms seems like a very short timeout to have?
> 
> /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 SamyGo
> Skickat: den 10 augusti 2012 10:11
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] DTMF detection giving problems
> 
> No Ivan , I've no issue at all with DTMFs ever. Like I said there is an Asterisk server which used torun the same IVRs and on same setup and never got such a problem.
> But not I'm relaying calls from Asterisk to FS and thus I've a SIP peer in between. Calls are coming in OK, I intermittently face this issue where user presses DTMFs and those digits are not recognized by FS-PlayAndGetDigits()
> I've  lines like this in my LUA
> 
>  choice = session:playAndGetDigits(1, 1,1,7, "#",sound_files[1], "", "[0-9]")
>  freeswitch.consoleLog("INFO","User Entered "..choice.." for this Survey\n")
> 
> Logs for the calls without dtmf here:
> 
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:1280
> 
> --------------------- Another Example---------------------
> 
> [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 0:960
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm
> 
> --------------------- Another Example---------------------
> 
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
> [INFO] switch_cpp.cpp:1227 User Entered  for this Survey
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 6:2560
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm
> 
> 
> 
> Log for successful DTMF collected is as follows:
> 
> [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed.
> [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:8800
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm
> [DEBUG] switch_ivr_play_say.c:2034 Test Regex [1][0-9]
> [INFO] switch_cpp.cpp:1227 User Entered 1 for this Survey
> [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
> [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm
> -------
> 
> 
> So I can see a pattern here. But I can't figure out this behavior. Is it because I've 1 try only ! and after just once playing of file PAGD() breaks no matter it gets DTMF or not !?
> 
> Regards,
> Sammy
> 
> 
> On Fri, Aug 10, 2012 at 10:09 AM, ivan baydin <runtwistedrat at gmail.com<mailto:runtwistedrat at gmail.com>> wrote:
> Hello Sammy.
> should check whether the operator is not blocking dtmf
> I had a situation - in the same statement it worked, on the other - not processed incoming dtmf
> 
> 2012/8/10 SamyGo <govoiper at gmail.com<mailto:govoiper at gmail.com>>
> Hello All,
> 
> I've a small lua script which is just an IVR. Calls are landing on Asterisk and then they are relayed over to FS. The problem is that the IVR don't always detects what user pressed. What I've observed is that the DTMF threshold is very high and some people just press slowly/quickly and the tone duration gets low and FS don't detect anything.
> 
> I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've also liberal-dtmf set in my sofia external.xml profile file which somewhat got me better results but still DTMFs miss sometimes.
> 
> Please note that the asterisk on which the calls are landing have never faced this issue before. Asterisk has PRI from which it receives the calls and then relay to desired FS server.
> 
> 
> Regards.
> Sammy
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
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> 
> 
> _________________________________________________________________________
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> 
> 
> 
> 
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> Join Us At ClueCon - Aug 7-9, 2012
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> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
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> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
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> 
> 
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> 
> 
> 
> 
> Official FreeSWITCH Sites
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> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
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> 
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-- 
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How do
you spend it?

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         covici at ccs.covici.com


From dujinfang at gmail.com  Fri Aug 10 17:24:30 2012
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 10 Aug 2012 21:24:30 +0800
Subject: [Freeswitch-users] Send event when FS starting
In-Reply-To: <CANJXrA8ehM++E63rdJ7qYz96NeHu-MOf3VqDrJmvJi95kTkzog@mail.gmail.com>
References: <CANJXrA8ehM++E63rdJ7qYz96NeHu-MOf3VqDrJmvJi95kTkzog@mail.gmail.com>
Message-ID: <3B5F528F0CA643259BF704B6130990FE@gmail.com>

I guess no, but you could look at the lua mod, it can run a lua script so you could original a channel and let outbound client know.

On Friday, August 10, 2012 at 4:36 PM, Prevail Magid wrote:

> Hello, 
> 
> Is it possible to configure event_socket to send event like 'FREESWITCH has been started' on outbound client? 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
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> 
> Join Us At ClueCon - Aug 7-9, 2012
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> 


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From grcamauer at gmail.com  Fri Aug 10 18:28:21 2012
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Fri, 10 Aug 2012 11:28:21 -0300
Subject: [Freeswitch-users] Error with MAKE CURRENT on SPANDSP
Message-ID: <CAPNTnnqGwGC=9uC1VH+=9xrtjHcskL9Hn6PP1hiyYCZWRnafCg@mail.gmail.com>

On CentOS 6.3, fully updated.  Was on FreeSWITCH Version
1.2.0-rc2+git~20120621T145213Z~2091f4f0d3+unclean~20120622T125056Z Started.
Just tried to "make current" and it choked when it got to SPANDSP:
...
making all mod_spandsp
make[5]: Entering directory
`/usr/src/freeswitch/src/mod/applications/mod_spandsp'
Creating mod_spandsp_la-mod_spandsp.lo
quiet_libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I../../../../src/include
-I../../../../libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include
-I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include
-I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
-DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1
-I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include
-I/usr/src/freeswitch/src/include
-I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement
-I/usr/src/freeswitch/libs/spandsp/src
-I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff
-I/usr/src/freeswitch/libs/spandsp/src
-I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT
mod_spandsp_la-mod_spandsp.lo -MD -MP -MF
.deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c  -fPIC -DPIC -o
.libs/mod_spandsp_la-mod_spandsp.o
In file included from
/usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:82,
                 from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
                 from mod_spandsp.h:50,
                 from mod_spandsp.c:39:
/usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:109: error:
expected specifier-qualifier-list before ?t81_t82_arith_encode_state_t?
/usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:202: error:
expected specifier-qualifier-list before ?t81_t82_arith_decode_state_t?
In file included from
/usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83,
                 from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
                 from mod_spandsp.h:50,
                 from mod_spandsp.c:39:
/usr/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54: error:
expected specifier-qualifier-list before ?lab_params_t?
/usr/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:77: error:
expected specifier-qualifier-list before ?lab_params_t?
In file included from
/usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:85,
                 from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
                 from mod_spandsp.h:50,
                 from mod_spandsp.c:39:
/usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_rx.h:112: error:
expected specifier-qualifier-list before ?t42_decode_state_t?
In file included from
/usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:86,
                 from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
                 from mod_spandsp.h:50,
                 from mod_spandsp.c:39:
/usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:136: error:
expected specifier-qualifier-list before ?t42_encode_state_t?
/usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:146: error:
expected specifier-qualifier-list before ?lab_params_t?
In file included from
/usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96,
                 from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
                 from mod_spandsp.h:50,
                 from mod_spandsp.c:39:
/usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:33:
error: expected specifier-qualifier-list before
?ademco_contactid_report_func_t?
cc1: warnings being treated as errors
/usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:48:
error: struct has no members
make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1
make[5]: Leaving directory
`/usr/src/freeswitch/src/mod/applications/mod_spandsp'
make[4]: *** [mod_spandsp-all] Error 1
make[4]: Leaving directory `/usr/src/freeswitch/src/mod'
make[3]: *** [all-recursive] Error 1
make[3]: Leaving directory `/usr/src/freeswitch/src'
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory `/usr/src/freeswitch'
make[1]: *** [all] Error 2
make[1]: Leaving directory `/usr/src/freeswitch'
make: *** [current] Error 2
[root at Freeswitch freeswitch]#


Any ideas?  Anybody else having this problem?
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From nickolayr at gmail.com  Fri Aug 10 18:50:46 2012
From: nickolayr at gmail.com (Nikolay Rogoshchenkov)
Date: Fri, 10 Aug 2012 10:50:46 -0400
Subject: [Freeswitch-users] Error with MAKE CURRENT on SPANDSP
In-Reply-To: <CAPNTnnqGwGC=9uC1VH+=9xrtjHcskL9Hn6PP1hiyYCZWRnafCg@mail.gmail.com>
References: <CAPNTnnqGwGC=9uC1VH+=9xrtjHcskL9Hn6PP1hiyYCZWRnafCg@mail.gmail.com>
Message-ID: <CAAQCs33DM44TaOy9i9UmErLfZm31Y_R2zDXOUB2udFmc3xeuhg@mail.gmail.com>

I think the best idea will be to post it in Jira<http://jira.freeswitch.org/>
.

--
Rogoshchenkov Nikolay


On Fri, Aug 10, 2012 at 10:28 AM, Guillermo Ruiz Camauer <
grcamauer at gmail.com> wrote:

>
> On CentOS 6.3, fully updated.  Was on FreeSWITCH Version
> 1.2.0-rc2+git~20120621T145213Z~2091f4f0d3+unclean~20120622T125056Z Started.
> Just tried to "make current" and it choked when it got to SPANDSP:
> ...
> making all mod_spandsp
> make[5]: Entering directory
> `/usr/src/freeswitch/src/mod/applications/mod_spandsp'
> Creating mod_spandsp_la-mod_spandsp.lo
> quiet_libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I../../../../src/include
> -I../../../../libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include
> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include
> -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1
> -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include
> -I/usr/src/freeswitch/src/include
> -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
> -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement
> -I/usr/src/freeswitch/libs/spandsp/src
> -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff
> -I/usr/src/freeswitch/libs/spandsp/src
> -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT
> mod_spandsp_la-mod_spandsp.lo -MD -MP -MF
> .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c  -fPIC -DPIC -o
> .libs/mod_spandsp_la-mod_spandsp.o
> In file included from
> /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:82,
>                  from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>                  from mod_spandsp.h:50,
>                  from mod_spandsp.c:39:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:109: error:
> expected specifier-qualifier-list before ?t81_t82_arith_encode_state_t?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:202: error:
> expected specifier-qualifier-list before ?t81_t82_arith_decode_state_t?
> In file included from
> /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83,
>                  from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>                  from mod_spandsp.h:50,
>                  from mod_spandsp.c:39:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54: error:
> expected specifier-qualifier-list before ?lab_params_t?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:77: error:
> expected specifier-qualifier-list before ?lab_params_t?
> In file included from
> /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:85,
>                  from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>                  from mod_spandsp.h:50,
>                  from mod_spandsp.c:39:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_rx.h:112: error:
> expected specifier-qualifier-list before ?t42_decode_state_t?
> In file included from
> /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:86,
>                  from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>                  from mod_spandsp.h:50,
>                  from mod_spandsp.c:39:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:136: error:
> expected specifier-qualifier-list before ?t42_encode_state_t?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:146: error:
> expected specifier-qualifier-list before ?lab_params_t?
> In file included from
> /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96,
>                  from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>                  from mod_spandsp.h:50,
>                  from mod_spandsp.c:39:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:33:
> error: expected specifier-qualifier-list before
> ?ademco_contactid_report_func_t?
> cc1: warnings being treated as errors
> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:48:
> error: struct has no members
> make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1
> make[5]: Leaving directory
> `/usr/src/freeswitch/src/mod/applications/mod_spandsp'
> make[4]: *** [mod_spandsp-all] Error 1
> make[4]: Leaving directory `/usr/src/freeswitch/src/mod'
> make[3]: *** [all-recursive] Error 1
> make[3]: Leaving directory `/usr/src/freeswitch/src'
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory `/usr/src/freeswitch'
> make[1]: *** [all] Error 2
> make[1]: Leaving directory `/usr/src/freeswitch'
> make: *** [current] Error 2
> [root at Freeswitch freeswitch]#
>
>
> Any ideas?  Anybody else having this problem?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From grcamauer at gmail.com  Fri Aug 10 19:22:18 2012
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Fri, 10 Aug 2012 12:22:18 -0300
Subject: [Freeswitch-users] Error with MAKE CURRENT on SPANDSP
In-Reply-To: <CAAQCs33DM44TaOy9i9UmErLfZm31Y_R2zDXOUB2udFmc3xeuhg@mail.gmail.com>
References: <CAPNTnnqGwGC=9uC1VH+=9xrtjHcskL9Hn6PP1hiyYCZWRnafCg@mail.gmail.com>
	<CAAQCs33DM44TaOy9i9UmErLfZm31Y_R2zDXOUB2udFmc3xeuhg@mail.gmail.com>
Message-ID: <CAPNTnnqmcqxC+zaF+ZUOu5vPvK9U35qVotGOStiBZAmBWvSWwA@mail.gmail.com>

Jira:
FreeSWITCH <http://jira.freeswitch.org/browse/FS>
FS-4515 <http://jira.freeswitch.org/browse/FS-4515>

On Fri, Aug 10, 2012 at 11:50 AM, Nikolay Rogoshchenkov <nickolayr at gmail.com
> wrote:

> I think the best idea will be to post it in Jira<http://jira.freeswitch.org/>
> .
>
> --
> Rogoshchenkov Nikolay
>
>
> On Fri, Aug 10, 2012 at 10:28 AM, Guillermo Ruiz Camauer <
> grcamauer at gmail.com> wrote:
>
>>
>> On CentOS 6.3, fully updated.  Was on FreeSWITCH Version
>> 1.2.0-rc2+git~20120621T145213Z~2091f4f0d3+unclean~20120622T125056Z Started.
>> Just tried to "make current" and it choked when it got to SPANDSP:
>> ...
>> making all mod_spandsp
>> make[5]: Entering directory
>> `/usr/src/freeswitch/src/mod/applications/mod_spandsp'
>> Creating mod_spandsp_la-mod_spandsp.lo
>> quiet_libtool: compile:  gcc -DHAVE_CONFIG_H -I.
>> -I../../../../src/include -I../../../../libs/xmlrpc-c
>> -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include
>> -I/usr/src/freeswitch/src/include
>> -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1
>> -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include
>> -I/usr/src/freeswitch/src/include
>> -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror
>> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
>> -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement
>> -I/usr/src/freeswitch/libs/spandsp/src
>> -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff
>> -I/usr/src/freeswitch/libs/spandsp/src
>> -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT
>> mod_spandsp_la-mod_spandsp.lo -MD -MP -MF
>> .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c  -fPIC -DPIC -o
>> .libs/mod_spandsp_la-mod_spandsp.o
>> In file included from
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:82,
>>                  from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>>                  from mod_spandsp.h:50,
>>                  from mod_spandsp.c:39:
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:109: error:
>> expected specifier-qualifier-list before ?t81_t82_arith_encode_state_t?
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:202: error:
>> expected specifier-qualifier-list before ?t81_t82_arith_decode_state_t?
>> In file included from
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83,
>>                  from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>>                  from mod_spandsp.h:50,
>>                  from mod_spandsp.c:39:
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54: error:
>> expected specifier-qualifier-list before ?lab_params_t?
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:77: error:
>> expected specifier-qualifier-list before ?lab_params_t?
>> In file included from
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:85,
>>                  from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>>                  from mod_spandsp.h:50,
>>                  from mod_spandsp.c:39:
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_rx.h:112: error:
>> expected specifier-qualifier-list before ?t42_decode_state_t?
>> In file included from
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:86,
>>                  from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>>                  from mod_spandsp.h:50,
>>                  from mod_spandsp.c:39:
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:136: error:
>> expected specifier-qualifier-list before ?t42_encode_state_t?
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:146: error:
>> expected specifier-qualifier-list before ?lab_params_t?
>> In file included from
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96,
>>                  from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>>                  from mod_spandsp.h:50,
>>                  from mod_spandsp.c:39:
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:33:
>> error: expected specifier-qualifier-list before
>> ?ademco_contactid_report_func_t?
>> cc1: warnings being treated as errors
>> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:48:
>> error: struct has no members
>> make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1
>> make[5]: Leaving directory
>> `/usr/src/freeswitch/src/mod/applications/mod_spandsp'
>> make[4]: *** [mod_spandsp-all] Error 1
>> make[4]: Leaving directory `/usr/src/freeswitch/src/mod'
>> make[3]: *** [all-recursive] Error 1
>> make[3]: Leaving directory `/usr/src/freeswitch/src'
>> make[2]: *** [all-recursive] Error 1
>> make[2]: Leaving directory `/usr/src/freeswitch'
>> make[1]: *** [all] Error 2
>> make[1]: Leaving directory `/usr/src/freeswitch'
>> make: *** [current] Error 2
>> [root at Freeswitch freeswitch]#
>>
>>
>> Any ideas?  Anybody else having this problem?
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Guillermo Ruiz Camauer
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From steveu at coppice.org  Fri Aug 10 19:24:16 2012
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 10 Aug 2012 23:24:16 +0800
Subject: [Freeswitch-users] Error with MAKE CURRENT on SPANDSP
In-Reply-To: <CAAQCs33DM44TaOy9i9UmErLfZm31Y_R2zDXOUB2udFmc3xeuhg@mail.gmail.com>
References: <CAPNTnnqGwGC=9uC1VH+=9xrtjHcskL9Hn6PP1hiyYCZWRnafCg@mail.gmail.com>
	<CAAQCs33DM44TaOy9i9UmErLfZm31Y_R2zDXOUB2udFmc3xeuhg@mail.gmail.com>
Message-ID: <502527A0.4090805@coppice.org>

If he opens a jira it will be immediately closed with "not a bug". "make 
spandsp-reconf" on the other hand would let him make some progress. :-)

Steve

On 08/10/2012 10:50 PM, Nikolay Rogoshchenkov wrote:
> I think the best idea will be to post it in Jira 
> <http://jira.freeswitch.org/>.
>
> --
> Rogoshchenkov Nikolay
>
>
> On Fri, Aug 10, 2012 at 10:28 AM, Guillermo Ruiz Camauer 
> <grcamauer at gmail.com <mailto:grcamauer at gmail.com>> wrote:
>
>
>     On CentOS 6.3, fully updated. Was on FreeSWITCH Version
>     1.2.0-rc2+git~20120621T145213Z~2091f4f0d3+unclean~20120622T125056Z
>     Started.
>     Just tried to "make current" and it choked when it got to SPANDSP:
>     ...
>     making all mod_spandsp
>     make[5]: Entering directory
>     `/usr/src/freeswitch/src/mod/applications/mod_spandsp'
>     Creating mod_spandsp_la-mod_spandsp.lo
>     quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I.
>     -I../../../../src/include -I../../../../libs/xmlrpc-c
>     -I/usr/src/freeswitch/libs/curl/include
>     -I/usr/src/freeswitch/src/include
>     -I/usr/src/freeswitch/src/include
>     -I/usr/src/freeswitch/libs/libteletone/src -fPIC
>     -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1
>     -I/usr/src/freeswitch/libs/curl/include
>     -I/usr/src/freeswitch/src/include
>     -I/usr/src/freeswitch/src/include
>     -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror
>     -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1
>     -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic
>     -Wdeclaration-after-statement
>     -I/usr/src/freeswitch/libs/spandsp/src
>     -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff
>     -I/usr/src/freeswitch/libs/spandsp/src
>     -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT
>     mod_spandsp_la-mod_spandsp.lo -MD -MP -MF
>     .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC
>     -o .libs/mod_spandsp_la-mod_spandsp.o
>     In file included from
>     /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:82,
>     from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>     from mod_spandsp.h:50,
>     from mod_spandsp.c:39:
>     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:109:
>     error: expected specifier-qualifier-list before
>     ?t81_t82_arith_encode_state_t?
>     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:202:
>     error: expected specifier-qualifier-list before
>     ?t81_t82_arith_decode_state_t?
>     In file included from
>     /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83,
>     from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>     from mod_spandsp.h:50,
>     from mod_spandsp.c:39:
>     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54:
>     error: expected specifier-qualifier-list before ?lab_params_t?
>     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:77:
>     error: expected specifier-qualifier-list before ?lab_params_t?
>     In file included from
>     /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:85,
>     from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>     from mod_spandsp.h:50,
>     from mod_spandsp.c:39:
>     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_rx.h:112:
>     error: expected specifier-qualifier-list before ?t42_decode_state_t?
>     In file included from
>     /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:86,
>     from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>     from mod_spandsp.h:50,
>     from mod_spandsp.c:39:
>     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:136:
>     error: expected specifier-qualifier-list before ?t42_encode_state_t?
>     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:146:
>     error: expected specifier-qualifier-list before ?lab_params_t?
>     In file included from
>     /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96,
>     from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
>     from mod_spandsp.h:50,
>     from mod_spandsp.c:39:
>     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:33:
>     error: expected specifier-qualifier-list before
>     ?ademco_contactid_report_func_t?
>     cc1: warnings being treated as errors
>     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:48:
>     error: struct has no members
>     make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1
>     make[5]: Leaving directory
>     `/usr/src/freeswitch/src/mod/applications/mod_spandsp'
>     make[4]: *** [mod_spandsp-all] Error 1
>     make[4]: Leaving directory `/usr/src/freeswitch/src/mod'
>     make[3]: *** [all-recursive] Error 1
>     make[3]: Leaving directory `/usr/src/freeswitch/src'
>     make[2]: *** [all-recursive] Error 1
>     make[2]: Leaving directory `/usr/src/freeswitch'
>     make[1]: *** [all] Error 2
>     make[1]: Leaving directory `/usr/src/freeswitch'
>     make: *** [current] Error 2
>     [root at Freeswitch freeswitch]#
>
>
>     Any ideas? Anybody else having this problem?
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     FreeSWITCH-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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From grcamauer at gmail.com  Fri Aug 10 19:38:09 2012
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Fri, 10 Aug 2012 12:38:09 -0300
Subject: [Freeswitch-users] Error with MAKE CURRENT on SPANDSP
In-Reply-To: <502527A0.4090805@coppice.org>
References: <CAPNTnnqGwGC=9uC1VH+=9xrtjHcskL9Hn6PP1hiyYCZWRnafCg@mail.gmail.com>
	<CAAQCs33DM44TaOy9i9UmErLfZm31Y_R2zDXOUB2udFmc3xeuhg@mail.gmail.com>
	<502527A0.4090805@coppice.org>
Message-ID: <CAPNTnnoynZ_jAxnQRViWayirc3q4_cVDhVy=K+kmHsVF_EvQrg@mail.gmail.com>

That did it!  Thanks.
Will close my own JIRA ticket.

Guillermo

On Fri, Aug 10, 2012 at 12:24 PM, Steve Underwood <steveu at coppice.org>wrote:

> If he opens a jira it will be immediately closed with "not a bug". "make
> spandsp-reconf" on the other hand would let him make some progress. :-)
>
> Steve
>
> On 08/10/2012 10:50 PM, Nikolay Rogoshchenkov wrote:
> > I think the best idea will be to post it in Jira
> > <http://jira.freeswitch.org/>.
> >
> > --
> > Rogoshchenkov Nikolay
> >
> >
> > On Fri, Aug 10, 2012 at 10:28 AM, Guillermo Ruiz Camauer
> > <grcamauer at gmail.com <mailto:grcamauer at gmail.com>> wrote:
> >
> >
> >     On CentOS 6.3, fully updated. Was on FreeSWITCH Version
> >     1.2.0-rc2+git~20120621T145213Z~2091f4f0d3+unclean~20120622T125056Z
> >     Started.
> >     Just tried to "make current" and it choked when it got to SPANDSP:
> >     ...
> >     making all mod_spandsp
> >     make[5]: Entering directory
> >     `/usr/src/freeswitch/src/mod/applications/mod_spandsp'
> >     Creating mod_spandsp_la-mod_spandsp.lo
> >     quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I.
> >     -I../../../../src/include -I../../../../libs/xmlrpc-c
> >     -I/usr/src/freeswitch/libs/curl/include
> >     -I/usr/src/freeswitch/src/include
> >     -I/usr/src/freeswitch/src/include
> >     -I/usr/src/freeswitch/libs/libteletone/src -fPIC
> >     -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1
> >     -I/usr/src/freeswitch/libs/curl/include
> >     -I/usr/src/freeswitch/src/include
> >     -I/usr/src/freeswitch/src/include
> >     -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror
> >     -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1
> >     -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic
> >     -Wdeclaration-after-statement
> >     -I/usr/src/freeswitch/libs/spandsp/src
> >     -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff
> >     -I/usr/src/freeswitch/libs/spandsp/src
> >     -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT
> >     mod_spandsp_la-mod_spandsp.lo -MD -MP -MF
> >     .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC
> >     -o .libs/mod_spandsp_la-mod_spandsp.o
> >     In file included from
> >     /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:82,
> >     from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
> >     from mod_spandsp.h:50,
> >     from mod_spandsp.c:39:
> >     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:109:
> >     error: expected specifier-qualifier-list before
> >     ?t81_t82_arith_encode_state_t?
> >     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:202:
> >     error: expected specifier-qualifier-list before
> >     ?t81_t82_arith_decode_state_t?
> >     In file included from
> >     /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83,
> >     from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
> >     from mod_spandsp.h:50,
> >     from mod_spandsp.c:39:
> >     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54:
> >     error: expected specifier-qualifier-list before ?lab_params_t?
> >     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:77:
> >     error: expected specifier-qualifier-list before ?lab_params_t?
> >     In file included from
> >     /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:85,
> >     from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
> >     from mod_spandsp.h:50,
> >     from mod_spandsp.c:39:
> >     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_rx.h:112:
> >     error: expected specifier-qualifier-list before ?t42_decode_state_t?
> >     In file included from
> >     /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:86,
> >     from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
> >     from mod_spandsp.h:50,
> >     from mod_spandsp.c:39:
> >     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:136:
> >     error: expected specifier-qualifier-list before ?t42_encode_state_t?
> >     /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:146:
> >     error: expected specifier-qualifier-list before ?lab_params_t?
> >     In file included from
> >     /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96,
> >     from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135,
> >     from mod_spandsp.h:50,
> >     from mod_spandsp.c:39:
> >
> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:33:
> >     error: expected specifier-qualifier-list before
> >     ?ademco_contactid_report_func_t?
> >     cc1: warnings being treated as errors
> >
> /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:48:
> >     error: struct has no members
> >     make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1
> >     make[5]: Leaving directory
> >     `/usr/src/freeswitch/src/mod/applications/mod_spandsp'
> >     make[4]: *** [mod_spandsp-all] Error 1
> >     make[4]: Leaving directory `/usr/src/freeswitch/src/mod'
> >     make[3]: *** [all-recursive] Error 1
> >     make[3]: Leaving directory `/usr/src/freeswitch/src'
> >     make[2]: *** [all-recursive] Error 1
> >     make[2]: Leaving directory `/usr/src/freeswitch'
> >     make[1]: *** [all] Error 2
> >     make[1]: Leaving directory `/usr/src/freeswitch'
> >     make: *** [current] Error 2
> >     [root at Freeswitch freeswitch]#
> >
> >
> >     Any ideas? Anybody else having this problem?
> >
> >
> >
> >
> _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com
> >
> >     
> >     
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org
> >     http://wiki.freeswitch.org
> >     http://www.cluecon.com
> >
> >     Join Us At ClueCon - Aug 7-9, 2012
> >
> >     FreeSWITCH-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
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Guillermo Ruiz Camauer
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From steveu at coppice.org  Fri Aug 10 19:53:14 2012
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 10 Aug 2012 23:53:14 +0800
Subject: [Freeswitch-users] high load with ~200 fax calls despite low
 CPU utilization and low IO wait
In-Reply-To: <CAE2A9TCP0waoCDyB6aEkZ=viD5yUzJmnJpzsgFTA1UXAC6431w@mail.gmail.com>
References: <CAE2A9TCP0waoCDyB6aEkZ=viD5yUzJmnJpzsgFTA1UXAC6431w@mail.gmail.com>
Message-ID: <50252E6A.1070700@coppice.org>

On 08/10/2012 04:55 AM, Adam Johnston wrote:
> Hi all,
>
> My FreeSWITCH setup works as follow: I have a service that launches 
> and monitors faxes (via event sockets) on one of a few FreeSWITCH 
> instances. These FreeSWITCH instances are running on dual-core, 2ghz 
> CentOS 5.8 and CentOS 6.0 VMs. The issue is that once I get to ~200 
> simultaneous faxes on any one FreeSWITCH VM the load average shoots up 
> (typically ~30, although I've seen it get much higher) and the VM 
> becomes sluggish and occasionally unusable.
>
> top and sar output confirms that CPU utilization is pretty low, as are 
> IO wait and RAM usage.
>
> I'm using a git head from 2012-02-06, although I see the same problem 
> if update to a more current head. On the most recent head I tried 
> (2012-08-06) the issue is worse, and crashes with a backtrace similar 
> to this Jira, http://jira.freeswitch.org/browse/FS-2893 despite my 
> compiling Lua and including the FreeSWITCH bindings.
>
> These issues occur on both the CentOS 5.8 and 6.0 VMs. I'm in the 
> process of loading 5.8 onto a physical machine and testing there. It's 
> also worth nothing that this load issue does not occur when I launch 
> faxes from a script on one of the VMs.
>
> Has anyone else seen similar high load issues before?
>
> Many thanks,
> Adam Johnston
>
What are these 200 FAXes doing? sending or receiving? Using audio or 
T.38? It makes quite a different to the load. Sending 200 faxes by audio 
at 14,400bps will keep a dual core machine busy. I'm not sure you can 
trust the CPU utilisation figures in a VM environment.

Steve



From ajohnston at blimessaging.com  Fri Aug 10 20:39:32 2012
From: ajohnston at blimessaging.com (Adam Johnston)
Date: Fri, 10 Aug 2012 12:39:32 -0400
Subject: [Freeswitch-users] high load with ~200 fax calls despite low
 CPU utilization and low IO wait
In-Reply-To: <50252E6A.1070700@coppice.org>
References: <CAE2A9TCP0waoCDyB6aEkZ=viD5yUzJmnJpzsgFTA1UXAC6431w@mail.gmail.com>
	<50252E6A.1070700@coppice.org>
Message-ID: <CAE2A9TDvqShX9Ku0uRowj=2QoudC=Ot7x3caU=M1G_eDf7UcnA@mail.gmail.com>

The 200 faxes are sending, they attempt T.38 but fallback to audio so it's
a mix of both.

If I log into a FreeSWITCH VM and launch 200 faxes from a script I don't
see nearly the same level of CPU load, so I'm inclined to think it has
something to do with the (many) event socket connections between my
controller application and the VMs. I'm setting up a hardware FreeSWITCH
server and writing a test harness to prove this out.

Thanks,
Adam


On Fri, Aug 10, 2012 at 11:53 AM, Steve Underwood <steveu at coppice.org>wrote:

> On 08/10/2012 04:55 AM, Adam Johnston wrote:
> > Hi all,
> >
> > My FreeSWITCH setup works as follow: I have a service that launches
> > and monitors faxes (via event sockets) on one of a few FreeSWITCH
> > instances. These FreeSWITCH instances are running on dual-core, 2ghz
> > CentOS 5.8 and CentOS 6.0 VMs. The issue is that once I get to ~200
> > simultaneous faxes on any one FreeSWITCH VM the load average shoots up
> > (typically ~30, although I've seen it get much higher) and the VM
> > becomes sluggish and occasionally unusable.
> >
> > top and sar output confirms that CPU utilization is pretty low, as are
> > IO wait and RAM usage.
> >
> > I'm using a git head from 2012-02-06, although I see the same problem
> > if update to a more current head. On the most recent head I tried
> > (2012-08-06) the issue is worse, and crashes with a backtrace similar
> > to this Jira, http://jira.freeswitch.org/browse/FS-2893 despite my
> > compiling Lua and including the FreeSWITCH bindings.
> >
> > These issues occur on both the CentOS 5.8 and 6.0 VMs. I'm in the
> > process of loading 5.8 onto a physical machine and testing there. It's
> > also worth nothing that this load issue does not occur when I launch
> > faxes from a script on one of the VMs.
> >
> > Has anyone else seen similar high load issues before?
> >
> > Many thanks,
> > Adam Johnston
> >
> What are these 200 FAXes doing? sending or receiving? Using audio or
> T.38? It makes quite a different to the load. Sending 200 faxes by audio
> at 14,400bps will keep a dual core machine busy. I'm not sure you can
> trust the CPU utilisation figures in a VM environment.
>
> Steve
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Aug 10 23:24:37 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 10 Aug 2012 14:24:37 -0500
Subject: [Freeswitch-users] high load with ~200 fax calls despite low
 CPU utilization and low IO wait
In-Reply-To: <CAE2A9TDvqShX9Ku0uRowj=2QoudC=Ot7x3caU=M1G_eDf7UcnA@mail.gmail.com>
References: <CAE2A9TCP0waoCDyB6aEkZ=viD5yUzJmnJpzsgFTA1UXAC6431w@mail.gmail.com>
	<50252E6A.1070700@coppice.org>
	<CAE2A9TDvqShX9Ku0uRowj=2QoudC=Ot7x3caU=M1G_eDf7UcnA@mail.gmail.com>
Message-ID: <CAKbxfG_+=exsSdhAN+YzMPbfJPi0Z4J0W3yBtbVtzB0xyg8vVg@mail.gmail.com>

Do not touch cent 6 with a 20 foot pole.   Something is seriously wrong
with it.  We have not figured it out yet.
On Aug 10, 2012 11:42 AM, "Adam Johnston" <ajohnston at blimessaging.com>
wrote:

> The 200 faxes are sending, they attempt T.38 but fallback to audio so it's
> a mix of both.
>
> If I log into a FreeSWITCH VM and launch 200 faxes from a script I don't
> see nearly the same level of CPU load, so I'm inclined to think it has
> something to do with the (many) event socket connections between my
> controller application and the VMs. I'm setting up a hardware FreeSWITCH
> server and writing a test harness to prove this out.
>
> Thanks,
> Adam
>
>
> On Fri, Aug 10, 2012 at 11:53 AM, Steve Underwood <steveu at coppice.org>wrote:
>
>> On 08/10/2012 04:55 AM, Adam Johnston wrote:
>> > Hi all,
>> >
>> > My FreeSWITCH setup works as follow: I have a service that launches
>> > and monitors faxes (via event sockets) on one of a few FreeSWITCH
>> > instances. These FreeSWITCH instances are running on dual-core, 2ghz
>> > CentOS 5.8 and CentOS 6.0 VMs. The issue is that once I get to ~200
>> > simultaneous faxes on any one FreeSWITCH VM the load average shoots up
>> > (typically ~30, although I've seen it get much higher) and the VM
>> > becomes sluggish and occasionally unusable.
>> >
>> > top and sar output confirms that CPU utilization is pretty low, as are
>> > IO wait and RAM usage.
>> >
>> > I'm using a git head from 2012-02-06, although I see the same problem
>> > if update to a more current head. On the most recent head I tried
>> > (2012-08-06) the issue is worse, and crashes with a backtrace similar
>> > to this Jira, http://jira.freeswitch.org/browse/FS-2893 despite my
>> > compiling Lua and including the FreeSWITCH bindings.
>> >
>> > These issues occur on both the CentOS 5.8 and 6.0 VMs. I'm in the
>> > process of loading 5.8 onto a physical machine and testing there. It's
>> > also worth nothing that this load issue does not occur when I launch
>> > faxes from a script on one of the VMs.
>> >
>> > Has anyone else seen similar high load issues before?
>> >
>> > Many thanks,
>> > Adam Johnston
>> >
>> What are these 200 FAXes doing? sending or receiving? Using audio or
>> T.38? It makes quite a different to the load. Sending 200 faxes by audio
>> at 14,400bps will keep a dual core machine busy. I'm not sure you can
>> trust the CPU utilisation figures in a VM environment.
>>
>> Steve
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mgg at giagnocavo.net  Fri Aug 10 23:33:07 2012
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Fri, 10 Aug 2012 19:33:07 +0000
Subject: [Freeswitch-users] high load with ~200 fax calls despite low
 CPU utilization and low IO wait
In-Reply-To: <CAKbxfG_+=exsSdhAN+YzMPbfJPi0Z4J0W3yBtbVtzB0xyg8vVg@mail.gmail.com>
References: <CAE2A9TCP0waoCDyB6aEkZ=viD5yUzJmnJpzsgFTA1UXAC6431w@mail.gmail.com>
	<50252E6A.1070700@coppice.org>
	<CAE2A9TDvqShX9Ku0uRowj=2QoudC=Ot7x3caU=M1G_eDf7UcnA@mail.gmail.com>
	<CAKbxfG_+=exsSdhAN+YzMPbfJPi0Z4J0W3yBtbVtzB0xyg8vVg@mail.gmail.com>
Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B612F933CD@BLUPRD0711MB413.namprd07.prod.outlook.com>

On CentOS 6.2 we see one thread that is runaway full CPU, but with 16 threads it isn't a high concern. Loading a few hundred channels doesn't seem to make the CPU go up very much. We haven't had any other issues with CentOS 6 in production for several months.

And yes, we plan on isolating it on a text machine and dumping and figuring out what is busted on that particular thread, just haven't had time to do it yet.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Friday, August 10, 2012 1:25 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] high load with ~200 fax calls despite low CPU utilization and low IO wait


Do not touch cent 6 with a 20 foot pole.   Something is seriously wrong with it.  We have not figured it out yet.
On Aug 10, 2012 11:42 AM, "Adam Johnston" <ajohnston at blimessaging.com<mailto:ajohnston at blimessaging.com>> wrote:
The 200 faxes are sending, they attempt T.38 but fallback to audio so it's a mix of both.

If I log into a FreeSWITCH VM and launch 200 faxes from a script I don't see nearly the same level of CPU load, so I'm inclined to think it has something to do with the (many) event socket connections between my controller application and the VMs. I'm setting up a hardware FreeSWITCH server and writing a test harness to prove this out.

Thanks,
Adam


On Fri, Aug 10, 2012 at 11:53 AM, Steve Underwood <steveu at coppice.org<mailto:steveu at coppice.org>> wrote:
On 08/10/2012 04:55 AM, Adam Johnston wrote:
> Hi all,
>
> My FreeSWITCH setup works as follow: I have a service that launches
> and monitors faxes (via event sockets) on one of a few FreeSWITCH
> instances. These FreeSWITCH instances are running on dual-core, 2ghz
> CentOS 5.8 and CentOS 6.0 VMs. The issue is that once I get to ~200
> simultaneous faxes on any one FreeSWITCH VM the load average shoots up
> (typically ~30, although I've seen it get much higher) and the VM
> becomes sluggish and occasionally unusable.
>
> top and sar output confirms that CPU utilization is pretty low, as are
> IO wait and RAM usage.
>
> I'm using a git head from 2012-02-06, although I see the same problem
> if update to a more current head. On the most recent head I tried
> (2012-08-06) the issue is worse, and crashes with a backtrace similar
> to this Jira, http://jira.freeswitch.org/browse/FS-2893 despite my
> compiling Lua and including the FreeSWITCH bindings.
>
> These issues occur on both the CentOS 5.8 and 6.0 VMs. I'm in the
> process of loading 5.8 onto a physical machine and testing there. It's
> also worth nothing that this load issue does not occur when I launch
> faxes from a script on one of the VMs.
>
> Has anyone else seen similar high load issues before?
>
> Many thanks,
> Adam Johnston
>
What are these 200 FAXes doing? sending or receiving? Using audio or
T.38? It makes quite a different to the load. Sending 200 faxes by audio
at 14,400bps will keep a dual core machine busy. I'm not sure you can
trust the CPU utilisation figures in a VM environment.

Steve


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From ajohnston at blimessaging.com  Sat Aug 11 00:48:21 2012
From: ajohnston at blimessaging.com (Adam Johnston)
Date: Fri, 10 Aug 2012 16:48:21 -0400
Subject: [Freeswitch-users] high load with ~200 fax calls despite low
 CPU utilization and low IO wait
In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B612F933CD@BLUPRD0711MB413.namprd07.prod.outlook.com>
References: <CAE2A9TCP0waoCDyB6aEkZ=viD5yUzJmnJpzsgFTA1UXAC6431w@mail.gmail.com>
	<50252E6A.1070700@coppice.org>
	<CAE2A9TDvqShX9Ku0uRowj=2QoudC=Ot7x3caU=M1G_eDf7UcnA@mail.gmail.com>
	<CAKbxfG_+=exsSdhAN+YzMPbfJPi0Z4J0W3yBtbVtzB0xyg8vVg@mail.gmail.com>
	<63B00DD1DA6A364E9F64A3A0BD2FE7B612F933CD@BLUPRD0711MB413.namprd07.prod.outlook.com>
Message-ID: <CAE2A9TCuh4_cusouPY5=DuF3V7hUGi0HdT4zWA8VHBgvjZvG=w@mail.gmail.com>

I have every intention of replacing our CentOS 6 servers with CentOS 5.8
once I get this load average issue properly understood. I would note that I
can reproduce this issue in both 6 and 5.8.

Adam


On Fri, Aug 10, 2012 at 3:33 PM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:

>  On CentOS 6.2 we see one thread that is runaway full CPU, but with 16
> threads it isn?t a high concern. Loading a few hundred channels doesn?t
> seem to make the CPU go up very much. We haven?t had any other issues with
> CentOS 6 in production for several months.****
>
> ** **
>
> And yes, we plan on isolating it on a text machine and dumping and
> figuring out what is busted on that particular thread, just haven?t had
> time to do it yet.****
>
> ** **
>
> -Michael****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Friday, August 10, 2012 1:25 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] high load with ~200 fax calls despite
> low CPU utilization and low IO wait****
>
> ** **
>
> Do not touch cent 6 with a 20 foot pole.   Something is seriously wrong
> with it.  We have not figured it out yet.****
>
> On Aug 10, 2012 11:42 AM, "Adam Johnston" <ajohnston at blimessaging.com>
> wrote:****
>
> The 200 faxes are sending, they attempt T.38 but fallback to audio so it's
> a mix of both.****
>
> ** **
>
> If I log into a FreeSWITCH VM and launch 200 faxes from a script I don't
> see nearly the same level of CPU load, so I'm inclined to think it has
> something to do with the (many) event socket connections between my
> controller application and the VMs. I'm setting up a hardware FreeSWITCH
> server and writing a test harness to prove this out.****
>
> ** **
>
> Thanks,****
>
> Adam****
>
> ** **
>
> ** **
>
> On Fri, Aug 10, 2012 at 11:53 AM, Steve Underwood <steveu at coppice.org>
> wrote:****
>
> On 08/10/2012 04:55 AM, Adam Johnston wrote:
> > Hi all,
> >
> > My FreeSWITCH setup works as follow: I have a service that launches
> > and monitors faxes (via event sockets) on one of a few FreeSWITCH
> > instances. These FreeSWITCH instances are running on dual-core, 2ghz
> > CentOS 5.8 and CentOS 6.0 VMs. The issue is that once I get to ~200
> > simultaneous faxes on any one FreeSWITCH VM the load average shoots up
> > (typically ~30, although I've seen it get much higher) and the VM
> > becomes sluggish and occasionally unusable.
> >
> > top and sar output confirms that CPU utilization is pretty low, as are
> > IO wait and RAM usage.
> >
> > I'm using a git head from 2012-02-06, although I see the same problem
> > if update to a more current head. On the most recent head I tried
> > (2012-08-06) the issue is worse, and crashes with a backtrace similar
> > to this Jira, http://jira.freeswitch.org/browse/FS-2893 despite my
> > compiling Lua and including the FreeSWITCH bindings.
> >
> > These issues occur on both the CentOS 5.8 and 6.0 VMs. I'm in the
> > process of loading 5.8 onto a physical machine and testing there. It's
> > also worth nothing that this load issue does not occur when I launch
> > faxes from a script on one of the VMs.
> >
> > Has anyone else seen similar high load issues before?
> >
> > Many thanks,
> > Adam Johnston
> >****
>
> What are these 200 FAXes doing? sending or receiving? Using audio or
> T.38? It makes quite a different to the load. Sending 200 faxes by audio
> at 14,400bps will keep a dual core machine busy. I'm not sure you can
> trust the CPU utilisation figures in a VM environment.
>
> Steve****
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.jj at googlemail.com  Fri Aug 10 12:43:14 2012
From: lists.jj at googlemail.com (Johannes Jakob)
Date: Fri, 10 Aug 2012 10:43:14 +0200
Subject: [Freeswitch-users] Call Diversion/CF (302 Moved Temporarily)
	not working
In-Reply-To: <CAO3-fvaKVtN6HewXzSGRFK-Ot9uCcRm_z1Dsf=yD7ETT6oQE3w@mail.gmail.com>
References: <CAO3-fvaKVtN6HewXzSGRFK-Ot9uCcRm_z1Dsf=yD7ETT6oQE3w@mail.gmail.com>
Message-ID: <92ABEA32FEAC47B6AA4CAD1AB90FE1B8@gmail.com>

Hi guys,

so really nobody got an idea about that one?
It's driving me nuts ;)

_Can_ this be some kind of dialplan fuckup?

Regards,

Johannes 

-- 
Johannes Jakob
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Tuesday, 7. August 2012 at 18:03, Johannes Jakob wrote:

> Hi,
> 
> My setup looks like this:
> 
> User A (1001), User B (1002) and User C (1003) are registered to
> 13.13.13.66 via OpenSIPs Proxy 222.222.222.222.
> User B activated unconditional call forwarding on his phone to User C.
> Now User A (or any external caller) calls User B.
> 
> Until last week User A would then be able to talk to User C - the
> expected behaviour.
> On Saturday, I updated the freeswitch from version 9fe08675a1 to
> d1c3f910a6 to get and test some of Steve's fax changes.
> Since this upgrade, Call-Forwarding / Call Diversion is broken on a
> level, that I don't understand.
> 
> 
> User A: INVITE 1002 to B's phone
> => User B: 302 Moved Temporarily (Contact: <sip:1003 at sip.isp.net (mailto:1003 at sip.isp.net);user=phone>)
> => FS: ACK
> => FS: INVITE 1003 to *B*'s phone!
> => User B: 404 Not Found
> => FS: ACK
> => User A gets signalled the 404 Not Found.
> 
> packet trace below.
> 
> 
> 
> 
> I'm aware of
> http://jira.freeswitch.org/browse/FS-724
> http://jira.freeswitch.org/browse/FS-821
> 
> but those didn't help me solve my issue:
> 
> internal.xml: <param name="apply-nat-acl" value="nat.auto"/>
> internal.xml: <param name="aggressive-nat-detection" value="false"/>
> 
> I added aggressive-nat-detection=false yesterday, before it wasn't in
> the config at all.
> I also tried removing the nat.auto acl temporarily, but that didn't help either.
> 
> 
> Yesterday afternoon I updated to git c3de9637af, did several
> bootstrappings, cleans and makes... no change.
> 
> So today I tried to downgrade to 9fe08675a1, the version I had running
> until Friday and that was working fine... well, it doesn't now. So I
> can't even _prove_ it was working before :(
> 
> 
> 
> Can somebody please point me in the right direction?
> 
> 
> 
> 
> Any hint in the right direction is much appreciated!
> 
> 
> 
> Thanks!
> 
> 
> 
> 
> 
> ======================================================================================
> 
> SIP Trace (tcpdump):
> 
> 
> 16:38:58.950321 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472
> E....[ .>..;^...
> .......D.fINVITE sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg SIP/2.0
> Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285>
> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0
> Via: SIP/2.0/UDP
> 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB
> Max-Forwards: 28
> From: "User A" <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;tag=3r8K7r5Hppe6N
> To: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>
> Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd
> CSeq: 31839105 INVITE
> Contact: <sip:mod_sofia at 13.13.13.66 (mailto:mod_sofia at 13.13.13.66):5060>
> User-Agent: FreeSWITCH
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, conference, presence, dialog, line-seize,
> call-info, sla, include-session-description, presence.winfo,
> message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 442
> P-Key-Flags: keys="3"
> X-AUTH-IP: 137.137.137.245
> X-FS-Support: update_display,send_info
> Remote-Party-ID: "User A"
> <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;party=calling;screen=yes;privacy=off
> 
> v=0
> o=FreeSWITCH 1344322582 1344322583 IN IP4 13.13.13.66
> s=FreeSWITCH
> c=IN IP4 13.13.13.66
> t=0 0
> m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101 13
> a=rtpmap:98 G7221/32000
> a=fmtp:98 bitrate=48000
> a=rtpmap:99 G7221/16000
> a=fmtp:99 bitrate=32000
> a=rtpmap:100 iLBC/8000
> a=fmtp:100 mode=30
> a=rtpmap:101 teleph
> 16:38:58.950328 IP 222.222.222.222 > 172.16.12.210: udp
> E....[..>.4.^...
> ..one-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> m=video 27936 RTP/AVP 34 102
> a=rtpmap:34 H263/90000
> a=rtpmap:102 H264/90000
> 
> 16:38:58.979148 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 623
> E..... at .@.
> .
> ..^........w8$SIP/2.0 302 Moved Temporarily
> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0
> Via: SIP/2.0/UDP
> 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB
> Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285>
> From: "User A" <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;tag=3r8K7r5Hppe6N
> To: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>;tag=mzey6xfxhz
> Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd
> CSeq: 31839105 INVITE
> Contact: <sip:1003 at sip.isp.net (mailto:1003 at sip.isp.net);user=phone>
> Diversion: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>;reason="unconditional"
> Content-Length: 0
> 
> 
> 16:38:58.979954 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 358
> .^.... at .>.
> .......n.fACK sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg SIP/2.0
> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0
> From: "User A" <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;tag=3r8K7r5Hppe6N
> Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd
> To: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>;tag=mzey6xfxhz
> CSeq: 31839105 ACK
> Max-Forwards: 70
> Content-Length: 0
> 
> 
> 16:38:58.985405 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472
> E....\ .>..:^...
> .......ZgIINVITE sip:1003 at 137.137.137.245 (mailto:1003 at 137.137.137.245):7390;user=phone SIP/2.0
> Record-Route: <sip:222.222.222.222;lr;ftag=411c9KpNKZ4rH;did=e81.20343ac4>
> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0
> Via: SIP/2.0/UDP
> 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ
> Max-Forwards: 28
> From: "User A" <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;tag=411c9KpNKZ4rH
> To: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>
> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd
> CSeq: 31839105 INVITE
> Contact: <sip:mod_sofia at 13.13.13.66 (mailto:mod_sofia at 13.13.13.66):5060>
> User-Agent: FreeSWITCH
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, conference, presence, dialog, line-seize,
> call-info, sla, include-session-description, presence.winfo,
> message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 466
> P-Key-Flags: keys="3"
> X-AUTH-IP: 137.137.137.245
> X-FS-Support: update_display,send_info
> Remote-Party-ID: "User A"
> <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;party=calling;screen=yes;privacy=off
> 
> v=0
> o=FreeSWITCH 1344322582 1344322584 IN IP4 13.13.13.66
> s=FreeSWITCH
> c=IN IP4 13.13.13.66
> t=0 0
> m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101
> a=rtpmap:98 G7221/32000
> a=fmtp:98 bitrate=48000
> a=rtpmap:99 G7221/16000
> a=fmtp:99 bitrate=32000
> a=rtpmap:100 iLBC/8000
> a=fmtp:100 mode=30
> a=rtpmap:101 telephone-e
> 16:38:58.985409 IP 222.222.222.222 > 172.16.12.210: udp
> E....\..>.4.^...
> ..vent/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> m=video 27936 RTP/AVP 34 102
> a=rtpmap:34 H263/90000
> a=rtpmap:102 H264/90000
> 
> 16:38:59.031218 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 551
> E..C.. at .@..7
> ..^......../..SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0
> Via: SIP/2.0/UDP
> 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ
> From: "User A" <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;tag=411c9KpNKZ4rH
> To: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>
> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd
> CSeq: 31839105 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> PRACK, MESSAGE, INFO
> Allow-Events: talk, hold, refer
> Supported: timer, 100rel, replaces
> Content-Length: 0
> 
> 
> 16:38:59.031922 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 341
> E..q.. at .>...^...
> .......].mACK sip:1003 at 137.137.137.245 (mailto:1003 at 137.137.137.245):7390;user=phone SIP/2.0
> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0
> From: "User A" <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;tag=411c9KpNKZ4rH
> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd
> To: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>
> CSeq: 31839105 ACK
> Max-Forwards: 70
> Content-Length: 0
> 
> 
> ====================================================================================== 




From itispip-qq at hotmail.com  Fri Aug 10 14:35:18 2012
From: itispip-qq at hotmail.com (=?gb2312?B?zfXA7Q==?=)
Date: Fri, 10 Aug 2012 18:35:18 +0800
Subject: [Freeswitch-users] Does chatplan support javascript?
Message-ID: <BLU169-W2039DAEFAF1A65AF380A7A8CB30@phx.gbl>


Hi FS Gurus,
I'm new to freeswitch. Trying to use javascript in chatplan, but get error code c:716 Invalid chat application interface [javascript]!
Same fucntion call in Lua is OK. So looking chatplan not support javascript?
If yes, any plan to add javascript to chatplan? Chatplan mainly dealing with text manipulation, Lua is a bit weak at string handling.
/rgds, Pip 		 	   		  
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From jakub.nadolski at dgt.com.pl  Fri Aug 10 15:58:13 2012
From: jakub.nadolski at dgt.com.pl (Jakub Nadolski)
Date: Fri, 10 Aug 2012 13:58:13 +0200
Subject: [Freeswitch-users] Is authorizing presence subscriptions possible?
Message-ID: <5024F755.8080003@dgt.com.pl>

Hello everyone,

I have a question about presence in mod sofia. I use Aastra phones and 
BLF function works fine. But now I need to authorize subcriptions. So I 
need configuration in which not all users can subscribe and receive 
state of the extension in NOTIFY messages. Is that possible and if so 
how can I do that?

Thanks in advance,
Kuba






From info at pripojtese.net  Fri Aug 10 17:18:01 2012
From: info at pripojtese.net (Jakub Tencl)
Date: Fri, 10 Aug 2012 14:18:01 +0100
Subject: [Freeswitch-users] script for generating bulk calls
In-Reply-To: <5025094E.4090802@healings.org.uk>
References: <5025094E.4090802@healings.org.uk>
Message-ID: <50250A09.5000409@pripojtese.net>



Hello,

i am quite new with freeswitch. Can anybody tell me if Freeswitch has
module for generating bulk calls and if not can you afford some template
script for it or just some pointers where i have to go?

Thank you

Best Regards

Jakub








From magnus.kelly at gmail.com  Sat Aug 11 02:43:58 2012
From: magnus.kelly at gmail.com (Magnus)
Date: Fri, 10 Aug 2012 23:43:58 +0100
Subject: [Freeswitch-users] Passing DTMF in Conference?
Message-ID: <060101cd7749$a1713000$e4539000$@gmail.com>

Hello all,

 

Is it possible to allow dtmf to flow when calls are in conference bridge?
For example if one auto connects two users to conference bridge, they are
placed into conference automatically, at this point one of the conference
users' needs to be able to pass a digit to send as dtmf to the other
conference end point device,  which they need to detect, alas it seems that
the dtmf from user 1 is not bridged to user 2 by default? DTMF used is
RF2833 Any tips or thoughts on this welcome.

 

Thanks

Magnus

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From mitch.capper at gmail.com  Sat Aug 11 05:00:45 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Fri, 10 Aug 2012 18:00:45 -0700
Subject: [Freeswitch-users] Passing DTMF in Conference?
In-Reply-To: <060101cd7749$a1713000$e4539000$@gmail.com>
References: <060101cd7749$a1713000$e4539000$@gmail.com>
Message-ID: <CALn233q-e+n9-ht7wfyHNG72jTmyo-Cr1UPnGFY0MUbMa3m0uw@mail.gmail.com>

Read the conference page on the wiki there is a param that does
exactly what you want for DTMF pass through.

~mithc

On Fri, Aug 10, 2012 at 3:43 PM, Magnus <magnus.kelly at gmail.com> wrote:
> Hello all,
>
>
>
> Is it possible to allow dtmf to flow when calls are in conference bridge?
> For example if one auto connects two users to conference bridge, they are
> placed into conference automatically, at this point one of the conference
> users? needs to be able to pass a digit to send as dtmf to the other
> conference end point device,  which they need to detect, alas it seems that
> the dtmf from user 1 is not bridged to user 2 by default? DTMF used is
> RF2833 Any tips or thoughts on this welcome.
>
>
>
> Thanks
>
> Magnus
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Sat Aug 11 05:12:44 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 10 Aug 2012 20:12:44 -0500
Subject: [Freeswitch-users] Does chatplan support javascript?
In-Reply-To: <BLU169-W2039DAEFAF1A65AF380A7A8CB30@phx.gbl>
References: <BLU169-W2039DAEFAF1A65AF380A7A8CB30@phx.gbl>
Message-ID: <CAKbxfG9aUwUth9MVj49MD3WWWj6=rfB9-DeZUq-26LsjgbdQ-w@mail.gmail.com>

its entirely possible but does not yet exist.
I pushed a patch that may enable it, but i did not have a way to test
it atm.  So you will have to try it for me and see what happens.

try git HEAD



On Fri, Aug 10, 2012 at 5:35 AM, ?? <itispip-qq at hotmail.com> wrote:
> Hi FS Gurus,
>
> I'm new to freeswitch. Trying to use javascript in chatplan, but get error
> code c:716 Invalid chat application interface [javascript]!
>
> Same fucntion call in Lua is OK. So looking chatplan not support javascript?
>
> If yes, any plan to add javascript to chatplan? Chatplan mainly dealing with
> text manipulation, Lua is a bit weak at string handling.
>
> /rgds, Pip
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mitch.capper at gmail.com  Sat Aug 11 05:18:27 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Fri, 10 Aug 2012 18:18:27 -0700
Subject: [Freeswitch-users] Call Diversion/CF (302 Moved Temporarily)
	not working
In-Reply-To: <92ABEA32FEAC47B6AA4CAD1AB90FE1B8@gmail.com>
References: <CAO3-fvaKVtN6HewXzSGRFK-Ot9uCcRm_z1Dsf=yD7ETT6oQE3w@mail.gmail.com>
	<92ABEA32FEAC47B6AA4CAD1AB90FE1B8@gmail.com>
Message-ID: <CALn233rA9qsPdprDLhd_42jae-Esi9Gn2vcvx1FxKQ40gEWeJg@mail.gmail.com>

Its been said a million times on this ML for bugs to be posted to
jira,  further read the wiki on using git bisect to track down the
exact revision it broke in, that will greatly increase the chance of
the bug being taken care of asap.

~mitch

On Fri, Aug 10, 2012 at 1:43 AM, Johannes Jakob <lists.jj at googlemail.com> wrote:
> Hi guys,
>
> so really nobody got an idea about that one?
> It's driving me nuts ;)
>
> _Can_ this be some kind of dialplan fuckup?
>
> Regards,
>
> Johannes
>
> --
> Johannes Jakob
> Sent with Sparrow (http://www.sparrowmailapp.com/?sig)
>
>
> On Tuesday, 7. August 2012 at 18:03, Johannes Jakob wrote:
>
>> Hi,
>>
>> My setup looks like this:
>>
>> User A (1001), User B (1002) and User C (1003) are registered to
>> 13.13.13.66 via OpenSIPs Proxy 222.222.222.222.
>> User B activated unconditional call forwarding on his phone to User C.
>> Now User A (or any external caller) calls User B.
>>
>> Until last week User A would then be able to talk to User C - the
>> expected behaviour.
>> On Saturday, I updated the freeswitch from version 9fe08675a1 to
>> d1c3f910a6 to get and test some of Steve's fax changes.
>> Since this upgrade, Call-Forwarding / Call Diversion is broken on a
>> level, that I don't understand.
>>
>>
>> User A: INVITE 1002 to B's phone
>> => User B: 302 Moved Temporarily (Contact: <sip:1003 at sip.isp.net (mailto:1003 at sip.isp.net);user=phone>)
>> => FS: ACK
>> => FS: INVITE 1003 to *B*'s phone!
>> => User B: 404 Not Found
>> => FS: ACK
>> => User A gets signalled the 404 Not Found.
>>
>> packet trace below.
>>
>>
>>
>>
>> I'm aware of
>> http://jira.freeswitch.org/browse/FS-724
>> http://jira.freeswitch.org/browse/FS-821
>>
>> but those didn't help me solve my issue:
>>
>> internal.xml: <param name="apply-nat-acl" value="nat.auto"/>
>> internal.xml: <param name="aggressive-nat-detection" value="false"/>
>>
>> I added aggressive-nat-detection=false yesterday, before it wasn't in
>> the config at all.
>> I also tried removing the nat.auto acl temporarily, but that didn't help either.
>>
>>
>> Yesterday afternoon I updated to git c3de9637af, did several
>> bootstrappings, cleans and makes... no change.
>>
>> So today I tried to downgrade to 9fe08675a1, the version I had running
>> until Friday and that was working fine... well, it doesn't now. So I
>> can't even _prove_ it was working before :(
>>
>>
>>
>> Can somebody please point me in the right direction?
>>
>>
>>
>>
>> Any hint in the right direction is much appreciated!
>>
>>
>>
>> Thanks!
>>
>>
>>
>>
>>
>> ======================================================================================
>>
>> SIP Trace (tcpdump):
>>
>>
>> 16:38:58.950321 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472
>> E....[ .>..;^...
>> .......D.fINVITE sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg SIP/2.0
>> Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285>
>> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0
>> Via: SIP/2.0/UDP
>> 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB
>> Max-Forwards: 28
>> From: "User A" <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;tag=3r8K7r5Hppe6N
>> To: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>
>> Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd
>> CSeq: 31839105 INVITE
>> Contact: <sip:mod_sofia at 13.13.13.66 (mailto:mod_sofia at 13.13.13.66):5060>
>> User-Agent: FreeSWITCH
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, hold, conference, presence, dialog, line-seize,
>> call-info, sla, include-session-description, presence.winfo,
>> message-summary, refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 442
>> P-Key-Flags: keys="3"
>> X-AUTH-IP: 137.137.137.245
>> X-FS-Support: update_display,send_info
>> Remote-Party-ID: "User A"
>> <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;party=calling;screen=yes;privacy=off
>>
>> v=0
>> o=FreeSWITCH 1344322582 1344322583 IN IP4 13.13.13.66
>> s=FreeSWITCH
>> c=IN IP4 13.13.13.66
>> t=0 0
>> m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101 13
>> a=rtpmap:98 G7221/32000
>> a=fmtp:98 bitrate=48000
>> a=rtpmap:99 G7221/16000
>> a=fmtp:99 bitrate=32000
>> a=rtpmap:100 iLBC/8000
>> a=fmtp:100 mode=30
>> a=rtpmap:101 teleph
>> 16:38:58.950328 IP 222.222.222.222 > 172.16.12.210: udp
>> E....[..>.4.^...
>> ..one-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> m=video 27936 RTP/AVP 34 102
>> a=rtpmap:34 H263/90000
>> a=rtpmap:102 H264/90000
>>
>> 16:38:58.979148 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 623
>> E..... at .@.
>> .
>> ..^........w8$SIP/2.0 302 Moved Temporarily
>> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0
>> Via: SIP/2.0/UDP
>> 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB
>> Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285>
>> From: "User A" <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;tag=3r8K7r5Hppe6N
>> To: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>;tag=mzey6xfxhz
>> Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd
>> CSeq: 31839105 INVITE
>> Contact: <sip:1003 at sip.isp.net (mailto:1003 at sip.isp.net);user=phone>
>> Diversion: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>;reason="unconditional"
>> Content-Length: 0
>>
>>
>> 16:38:58.979954 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 358
>> .^.... at .>.
>> .......n.fACK sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg SIP/2.0
>> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0
>> From: "User A" <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;tag=3r8K7r5Hppe6N
>> Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd
>> To: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>;tag=mzey6xfxhz
>> CSeq: 31839105 ACK
>> Max-Forwards: 70
>> Content-Length: 0
>>
>>
>> 16:38:58.985405 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472
>> E....\ .>..:^...
>> .......ZgIINVITE sip:1003 at 137.137.137.245 (mailto:1003 at 137.137.137.245):7390;user=phone SIP/2.0
>> Record-Route: <sip:222.222.222.222;lr;ftag=411c9KpNKZ4rH;did=e81.20343ac4>
>> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0
>> Via: SIP/2.0/UDP
>> 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ
>> Max-Forwards: 28
>> From: "User A" <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;tag=411c9KpNKZ4rH
>> To: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>
>> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd
>> CSeq: 31839105 INVITE
>> Contact: <sip:mod_sofia at 13.13.13.66 (mailto:mod_sofia at 13.13.13.66):5060>
>> User-Agent: FreeSWITCH
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, hold, conference, presence, dialog, line-seize,
>> call-info, sla, include-session-description, presence.winfo,
>> message-summary, refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 466
>> P-Key-Flags: keys="3"
>> X-AUTH-IP: 137.137.137.245
>> X-FS-Support: update_display,send_info
>> Remote-Party-ID: "User A"
>> <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;party=calling;screen=yes;privacy=off
>>
>> v=0
>> o=FreeSWITCH 1344322582 1344322584 IN IP4 13.13.13.66
>> s=FreeSWITCH
>> c=IN IP4 13.13.13.66
>> t=0 0
>> m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101
>> a=rtpmap:98 G7221/32000
>> a=fmtp:98 bitrate=48000
>> a=rtpmap:99 G7221/16000
>> a=fmtp:99 bitrate=32000
>> a=rtpmap:100 iLBC/8000
>> a=fmtp:100 mode=30
>> a=rtpmap:101 telephone-e
>> 16:38:58.985409 IP 222.222.222.222 > 172.16.12.210: udp
>> E....\..>.4.^...
>> ..vent/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> m=video 27936 RTP/AVP 34 102
>> a=rtpmap:34 H263/90000
>> a=rtpmap:102 H264/90000
>>
>> 16:38:59.031218 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 551
>> E..C.. at .@..7
>> ..^......../..SIP/2.0 404 Not Found
>> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0
>> Via: SIP/2.0/UDP
>> 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ
>> From: "User A" <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;tag=411c9KpNKZ4rH
>> To: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>
>> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd
>> CSeq: 31839105 INVITE
>> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
>> PRACK, MESSAGE, INFO
>> Allow-Events: talk, hold, refer
>> Supported: timer, 100rel, replaces
>> Content-Length: 0
>>
>>
>> 16:38:59.031922 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 341
>> E..q.. at .>...^...
>> .......].mACK sip:1003 at 137.137.137.245 (mailto:1003 at 137.137.137.245):7390;user=phone SIP/2.0
>> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0
>> From: "User A" <sip:1001 at sip.isp.net (mailto:1001 at sip.isp.net)>;tag=411c9KpNKZ4rH
>> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd
>> To: <sip:1002 at 137.137.137.245 (mailto:1002 at 137.137.137.245):7390;line=ydwzxzpg>
>> CSeq: 31839105 ACK
>> Max-Forwards: 70
>> Content-Length: 0
>>
>>
>> ======================================================================================
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From krice at freeswitch.org  Sat Aug 11 05:19:33 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 10 Aug 2012 20:19:33 -0500
Subject: [Freeswitch-users] Is authorizing presence subscriptions
	possible?
In-Reply-To: <5024F755.8080003@dgt.com.pl>
References: <5024F755.8080003@dgt.com.pl>
Message-ID: <D869AABB-A75F-4198-8638-B7E06287C146@freeswitch.org>


as in on a per user basis? ie user x can subscribe but user y cannot?
Ken
Sent from my iPad

On Aug 10, 2012, at 6:58 AM, Jakub Nadolski <jakub.nadolski at dgt.com.pl> wrote:

> Hello everyone,
> 
> I have a question about presence in mod sofia. I use Aastra phones and 
> BLF function works fine. But now I need to authorize subcriptions. So I 
> need configuration in which not all users can subscribe and receive 
> state of the extension in NOTIFY messages. Is that possible and if so 
> how can I do that?
> 
> Thanks in advance,
> Kuba
> 
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Sat Aug 11 05:30:59 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 10 Aug 2012 20:30:59 -0500
Subject: [Freeswitch-users] high load with ~200 fax calls despite low
 CPU utilization and low IO wait
In-Reply-To: <CAE2A9TCuh4_cusouPY5=DuF3V7hUGi0HdT4zWA8VHBgvjZvG=w@mail.gmail.com>
References: <CAE2A9TCP0waoCDyB6aEkZ=viD5yUzJmnJpzsgFTA1UXAC6431w@mail.gmail.com>
	<50252E6A.1070700@coppice.org>
	<CAE2A9TDvqShX9Ku0uRowj=2QoudC=Ot7x3caU=M1G_eDf7UcnA@mail.gmail.com>
	<CAKbxfG_+=exsSdhAN+YzMPbfJPi0Z4J0W3yBtbVtzB0xyg8vVg@mail.gmail.com>
	<63B00DD1DA6A364E9F64A3A0BD2FE7B612F933CD@BLUPRD0711MB413.namprd07.prod.outlook.com>
	<CAE2A9TCuh4_cusouPY5=DuF3V7hUGi0HdT4zWA8VHBgvjZvG=w@mail.gmail.com>
Message-ID: <CAKbxfG_CyQqbjn99xi3fj_F_KsC0DxF+Cp-tX1y8_K3Ud4mDhw@mail.gmail.com>

If you consider this an issue.... Move it to Jira.
 On Aug 10, 2012 3:50 PM, "Adam Johnston" <ajohnston at blimessaging.com>
wrote:

> I have every intention of replacing our CentOS 6 servers with CentOS 5.8
> once I get this load average issue properly understood. I would note that I
> can reproduce this issue in both 6 and 5.8.
>
> Adam
>
>
> On Fri, Aug 10, 2012 at 3:33 PM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:
>
>>  On CentOS 6.2 we see one thread that is runaway full CPU, but with 16
>> threads it isn?t a high concern. Loading a few hundred channels doesn?t
>> seem to make the CPU go up very much. We haven?t had any other issues with
>> CentOS 6 in production for several months.****
>>
>> ** **
>>
>> And yes, we plan on isolating it on a text machine and dumping and
>> figuring out what is busted on that particular thread, just haven?t had
>> time to do it yet.****
>>
>> ** **
>>
>> -Michael****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
>> Minessale
>> *Sent:* Friday, August 10, 2012 1:25 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] high load with ~200 fax calls despite
>> low CPU utilization and low IO wait****
>>
>> ** **
>>
>> Do not touch cent 6 with a 20 foot pole.   Something is seriously wrong
>> with it.  We have not figured it out yet.****
>>
>> On Aug 10, 2012 11:42 AM, "Adam Johnston" <ajohnston at blimessaging.com>
>> wrote:****
>>
>> The 200 faxes are sending, they attempt T.38 but fallback to audio so
>> it's a mix of both.****
>>
>> ** **
>>
>> If I log into a FreeSWITCH VM and launch 200 faxes from a script I don't
>> see nearly the same level of CPU load, so I'm inclined to think it has
>> something to do with the (many) event socket connections between my
>> controller application and the VMs. I'm setting up a hardware FreeSWITCH
>> server and writing a test harness to prove this out.****
>>
>> ** **
>>
>> Thanks,****
>>
>> Adam****
>>
>> ** **
>>
>> ** **
>>
>> On Fri, Aug 10, 2012 at 11:53 AM, Steve Underwood <steveu at coppice.org>
>> wrote:****
>>
>> On 08/10/2012 04:55 AM, Adam Johnston wrote:
>> > Hi all,
>> >
>> > My FreeSWITCH setup works as follow: I have a service that launches
>> > and monitors faxes (via event sockets) on one of a few FreeSWITCH
>> > instances. These FreeSWITCH instances are running on dual-core, 2ghz
>> > CentOS 5.8 and CentOS 6.0 VMs. The issue is that once I get to ~200
>> > simultaneous faxes on any one FreeSWITCH VM the load average shoots up
>> > (typically ~30, although I've seen it get much higher) and the VM
>> > becomes sluggish and occasionally unusable.
>> >
>> > top and sar output confirms that CPU utilization is pretty low, as are
>> > IO wait and RAM usage.
>> >
>> > I'm using a git head from 2012-02-06, although I see the same problem
>> > if update to a more current head. On the most recent head I tried
>> > (2012-08-06) the issue is worse, and crashes with a backtrace similar
>> > to this Jira, http://jira.freeswitch.org/browse/FS-2893 despite my
>> > compiling Lua and including the FreeSWITCH bindings.
>> >
>> > These issues occur on both the CentOS 5.8 and 6.0 VMs. I'm in the
>> > process of loading 5.8 onto a physical machine and testing there. It's
>> > also worth nothing that this load issue does not occur when I launch
>> > faxes from a script on one of the VMs.
>> >
>> > Has anyone else seen similar high load issues before?
>> >
>> > Many thanks,
>> > Adam Johnston
>> >****
>>
>> What are these 200 FAXes doing? sending or receiving? Using audio or
>> T.38? It makes quite a different to the load. Sending 200 faxes by audio
>> at 14,400bps will keep a dual core machine busy. I'm not sure you can
>> trust the CPU utilisation figures in a VM environment.
>>
>> Steve****
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>> ** **
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Sat Aug 11 05:31:10 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 10 Aug 2012 21:31:10 -0400
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <50236299.4070403@softnet.si>
References: <1FFF97C269757C458224B7C895F35F15141094@cantor.std.visionutv.se>
	<50236299.4070403@softnet.si>
Message-ID: <CAKV-jFyJEGW72UwZ8AGkKPL9uvoab3fOUb7y5_wTx6_suPT0JQ@mail.gmail.com>

I need to update the wiki, but the install script for Debian is at
github.com/bdfoster/fs-debian-installer. It still gets the latest code on
master. If you guys want to give me some feedback I can incorporate the
proper changes. I stopped working on it because most are using centos and I
am not going to maintain something no one uses. I have the time, I have the
energy, but I need to know what people would like to see. Our internal
script is based on this one but I cannot release those changes to the world.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 9, 2012 3:12 AM, "Miha" <miha at softnet.si> wrote:

>  On 8/9/2012 8:36 AM, Peter Olsson wrote:
>
>  If you change to branch ?v1.2.stable?, the version will be 1.2.0.****
>
> ** **
>
> The core devs have not yet sent out any official instructions, but I guess
> that if you want to continue to follow the main development (with sometimes
> unstable code), just continue to do as you do today. If you want to stick
> to the 1.2 version, and only get bugfixes etc. for that version (but no new
> features), change to the branch v1.2.stable.****
>
> ** **
>
> I think there will be more official instructions soon, but this is that
> way I think it?s meant to work.****
>
> ** **
>
> /Peter****
>
> ** **
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
> *F?r *Miha
> *Skickat:* den 9 augusti 2012 08:15
> *Till:* FreeSWITCH Users Help
> *?mne:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2****
>
> ** **
>
> Hi,
>
> so my current version is 1.2 rc2. After make current I will have 1.2? I
> try it on test server but still FreeSWITCH Version
> 1.2.0-rc2+git~20120809T022906Z~3389f32363 (1.2.0-rc2; git at commit
> 3389f32363 on Thu, 09 Aug 2012 02:29:06 Z)
>
>
> Thanks!
>
> Miha
>
> On 8/8/2012 5:02 PM, Ken Rice wrote:****
>
> the Primary repo is git.freeswitch.org/freeswitch.git this is the master
> repo, however, the github.com/freeswitch/freeswitch is an unofficial
> mirror...  updates to primary git repo should be pushed including tags and
> branches but we do not support those that wish to use other than official
> repos ****
>
> ** **
>
> ** **
>
> K****
>
> ** **
>
> On Wed, Aug 8, 2012 at 8:39 AM, Brian Foster <bdfoster at endigotech.com>
> wrote:****
>
> Ah, looks like they publish to github as well. Afaik the primary repo is
> git.freeswitch.org, and that is where I get everything. ****
>
> Brian Foster
> Endigo Computer LLC****
>
> Sent from a mobile device.****
>
> On Aug 8, 2012 8:41 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:****
>
> What is this then? https://github.com/freeswitch/freeswitch ****
>
>
> Regards,
> Ben Langfeld
>
> ****
>
> On 8 August 2012 12:53, Brian Foster <bdfoster at endigotech.com> wrote:****
>
> FreeSWITCH isn't published on GitHub.****
>
> Brian Foster
> Endigo Computer LLC****
>
> Sent from a mobile device.****
>
> On Aug 8, 2012 3:14 AM, "Stanislav Sinyagin" <ssinyagin at yahoo.com> wrote:*
> ***
>
> Ken,****
>
> care to push tags to Github as well?****
>
> ** **
>   ------------------------------
>
> *From:* Ken Rice <krice at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 7, 2012 8:31 PM
> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2****
>
> ** **
>
> yes its there... git pull --tags****
>
> ** **
>
> On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net> wrote:****
>
> I also checked tags, but didn't see it. Are you sure your branch is
> tracked upstream?****
>
>  ****
>
> --Dave****
>
> ** **
>  ------------------------------
>
> *From:* Ken Rice [mailto:krice at freeswitch.org]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> ****
>
> *Sent:* Tue, 07 Aug 2012 11:15:50 -0700
> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2 ****
>
>
>
> its Tagged, right now, we will be branching it shortly for tracking
> patches. Please note we had a couple of good users point out some smallish
> bugs so we're int he process of fixing those! so guys if you find anything
> wrong please open a jira, if you are at cluecon, find me and let me know
> the bug number ****
>
> ** **
>
> K****
>
> On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net> wrote:****
>
> Where is the branch so I can checkout local?****
>
>  ****
>
> D:\fsnew>git branch -a
> * master
>   remotes/origin/FS-3432
>   remotes/origin/FS-4062
>   remotes/origin/HEAD -> origin/master
>   remotes/origin/dingaling_video
>   remotes/origin/master
>   remotes/origin/smgfs
>   remotes/origin/stable-test/freeswitch-1.2
>   remotes/origin/swk/fs_test
>   remotes/test/master****
>
> --Dave****
>
> ** **
>  ------------------------------
>
> *From:* Ken Rice [mailto:krice at freeswitch.org]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org],
> freeswitch-dev at lists.freeswitch.org
> *Sent:* Tue, 07 Aug 2012 09:03:34 -0700 ****
>
>
> *Subject:* [Freeswitch-users] Announcing FreeSWITCH 1.2****
>
> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0! ** **
>
> ** **
>
> Get your copy today at
> http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 ! ****
>
> ** **
>
> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
> continue to fix bugs and security issues. giving you a stable platform for
> at least one year.****
>
> ** **
>
> Grab it today!****
>
> ** **
>
> The FreeSWITCH Team ****
>
>   ****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com/
>
> 
> /
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org/
> http://wiki.freeswitch.org/
> http://www.cluecon.com/
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
>   ****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
>
>
>
> ****
>
> _________________________________________________________________________****
>
> Professional FreeSWITCH Consulting Services:****
>
> consulting at freeswitch.org****
>
> http://www.freeswitchsolutions.com****
>
> ** **
>
> ****
>
> ****
>
> ** **
>
> Official FreeSWITCH Sites****
>
> http://www.freeswitch.org****
>
> http://wiki.freeswitch.org****
>
> http://www.cluecon.com****
>
> ** **
>
> Join Us At ClueCon - Aug 7-9, 2012****
>
> ** **
>
> FreeSWITCH-users mailing list****
>
> FreeSWITCH-users at 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:502354a432765721010175! ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>  @Peter,
>
> thanks for all your explanation.
>
> Br,
> Miha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Sat Aug 11 05:43:50 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 10 Aug 2012 21:43:50 -0400
Subject: [Freeswitch-users] script for generating bulk calls
In-Reply-To: <50250A09.5000409@pripojtese.net>
References: <5025094E.4090802@healings.org.uk>
	<50250A09.5000409@pripojtese.net>
Message-ID: <CAKV-jFzWQ1U6f2f3zEP0mjdO3MvHW4f5u860q4LfoYhUZLARGA@mail.gmail.com>

Newfies Dialer or have it contracted out. We've done something similar
here.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 10, 2012 5:45 PM, "Jakub Tencl" <info at pripojtese.net> wrote:

>
>
> Hello,
>
> i am quite new with freeswitch. Can anybody tell me if Freeswitch has
> module for generating bulk calls and if not can you afford some template
> script for it or just some pointers where i have to go?
>
> Thank you
>
> Best Regards
>
> Jakub
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vishal.kakkar at gmail.com  Sat Aug 11 07:25:12 2012
From: vishal.kakkar at gmail.com (Vishal Kakkar)
Date: Sat, 11 Aug 2012 08:55:12 +0530
Subject: [Freeswitch-users] Collateral Damage
Message-ID: <CAAxwcqvqEUoNnbG+kiPJeo0Ohx20HVtE0oz6GyjLyJrF7uB4cQ@mail.gmail.com>

While doing FTP(Upload) my public.xml to server machine, internet
connection broken and now my public.xml is empty(0 bytes).
But my Freeswitch is still running with last file loaded.

Is there any way i can get my public.xml which is currently loaded in
running freeswitch.

Please help guys..
Thanks a lot
-Vishu
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From chad at apartmentlines.com  Sat Aug 11 07:44:34 2012
From: chad at apartmentlines.com (Chad Phillips)
Date: Fri, 10 Aug 2012 20:44:34 -0700
Subject: [Freeswitch-users] Collateral Damage
In-Reply-To: <CAAxwcqvqEUoNnbG+kiPJeo0Ohx20HVtE0oz6GyjLyJrF7uB4cQ@mail.gmail.com>
References: <CAAxwcqvqEUoNnbG+kiPJeo0Ohx20HVtE0oz6GyjLyJrF7uB4cQ@mail.gmail.com>
Message-ID: <960588CE0EB9415696CF8173741807C6@gmail.com>

pretty sure log/freeswitch.xml.fsxml has a complete copy of the XML that's loaded? 

chad 


On Friday, August 10, 2012 at 8:25 PM, Vishal Kakkar wrote:

> While doing FTP(Upload) my public.xml to server machine, internet connection broken and now my public.xml is empty(0 bytes).
> But my Freeswitch is still running with last file loaded.
> 
> Is there any way i can get my public.xml which is currently loaded in running freeswitch.
> 
> Please help guys.. 
> Thanks a lot
> -Vishu
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-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 vishal.kakkar at gmail.com  Sat Aug 11 07:56:45 2012
From: vishal.kakkar at gmail.com (Vishal Kakkar)
Date: Sat, 11 Aug 2012 09:26:45 +0530
Subject: [Freeswitch-users] Collateral Damage
In-Reply-To: <960588CE0EB9415696CF8173741807C6@gmail.com>
References: <CAAxwcqvqEUoNnbG+kiPJeo0Ohx20HVtE0oz6GyjLyJrF7uB4cQ@mail.gmail.com>
	<960588CE0EB9415696CF8173741807C6@gmail.com>
Message-ID: <CAAxwcqsPcL7yKXfbo5VY7ZYMXn2xF3h5vya=2T7DhiT+OXU+og@mail.gmail.com>

Thanks Chad.. Gr8 Help!!

On Sat, Aug 11, 2012 at 9:14 AM, Chad Phillips <chad at apartmentlines.com>wrote:

>  pretty sure log/freeswitch.xml.fsxml has a complete copy of the XML
> that's loaded?
>
> chad
>
> On Friday, August 10, 2012 at 8:25 PM, Vishal Kakkar wrote:
>
> While doing FTP(Upload) my public.xml to server machine, internet
> connection broken and now my public.xml is empty(0 bytes).
> But my Freeswitch is still running with last file loaded.
>
> Is there any way i can get my public.xml which is currently loaded in
> running freeswitch.
>
> Please help guys..
> Thanks a lot
> -Vishu
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vishal.kakkar at gmail.com  Sat Aug 11 08:53:08 2012
From: vishal.kakkar at gmail.com (Vishal Kakkar)
Date: Sat, 11 Aug 2012 10:23:08 +0530
Subject: [Freeswitch-users] mod_fifo
Message-ID: <CAAxwcquspdkRcNRfG9LA7kbie7Du3EKtFUvon=-6-24QLs5m=Q@mail.gmail.com>

Hi all,

I want to put the caller into fifo queue *without answering the call till
agent attend it.*
But fifo is answering the call while putting caller in queue.

Please help!!

Thanks,
-Vishal.
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From asaad2 at gmail.com  Sat Aug 11 11:41:58 2012
From: asaad2 at gmail.com (BookBag)
Date: Sat, 11 Aug 2012 02:41:58 -0500
Subject: [Freeswitch-users] script for generating bulk calls
In-Reply-To: <CAKV-jFzWQ1U6f2f3zEP0mjdO3MvHW4f5u860q4LfoYhUZLARGA@mail.gmail.com>
References: <5025094E.4090802@healings.org.uk>
	<50250A09.5000409@pripojtese.net>
	<CAKV-jFzWQ1U6f2f3zEP0mjdO3MvHW4f5u860q4LfoYhUZLARGA@mail.gmail.com>
Message-ID: <CAAdTMa-NqDg6rk-7+9QPJ7ZGTsQJcmApyd72WzmgY6MCq1TrOw@mail.gmail.com>

I had this problem and ended up first creating a script of a whole bunch
originate statements. Then  I found fusionpbx. It has a call broadcast
script already built it using a web interface. You might wanna take a look
at it.
On Aug 10, 2012 9:47 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> Newfies Dialer or have it contracted out. We've done something similar
> here.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Aug 10, 2012 5:45 PM, "Jakub Tencl" <info at pripojtese.net> wrote:
>
>>
>>
>> Hello,
>>
>> i am quite new with freeswitch. Can anybody tell me if Freeswitch has
>> module for generating bulk calls and if not can you afford some template
>> script for it or just some pointers where i have to go?
>>
>> Thank you
>>
>> Best Regards
>>
>> Jakub
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jakub.nadolski at dgt.com.pl  Sat Aug 11 11:54:21 2012
From: jakub.nadolski at dgt.com.pl (Jakub Nadolski)
Date: Sat, 11 Aug 2012 09:54:21 +0200 (CEST)
Subject: [Freeswitch-users] Is authorizing presence subscriptions
 possible?
In-Reply-To: <D869AABB-A75F-4198-8638-B7E06287C146@freeswitch.org>
References: <5024F755.8080003@dgt.com.pl>
	<D869AABB-A75F-4198-8638-B7E06287C146@freeswitch.org>
Message-ID: <52148.193.34.0.63.1344671661.squirrel@mail.dgtronik.com.pl>

Yes, this is what I would like to do.

Kuba

>
> as in on a per user basis? ie user x can subscribe but user y cannot?
> Ken
> Sent from my iPad
>
> On Aug 10, 2012, at 6:58 AM, Jakub Nadolski <jakub.nadolski at dgt.com.pl>
> wrote:
>
>> Hello everyone,
>>
>> I have a question about presence in mod sofia. I use Aastra phones and
>> BLF function works fine. But now I need to authorize subcriptions. So I
>> need configuration in which not all users can subscribe and receive
>> state of the extension in NOTIFY messages. Is that possible and if so
>> how can I do that?
>>
>> Thanks in advance,
>> Kuba
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From sos at sokhapkin.dyndns.org  Sat Aug 11 14:26:05 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 11 Aug 2012 06:26:05 -0400
Subject: [Freeswitch-users] mod_fifo
In-Reply-To: <CAAxwcquspdkRcNRfG9LA7kbie7Du3EKtFUvon=-6-24QLs5m=Q@mail.gmail.com>
References: <CAAxwcquspdkRcNRfG9LA7kbie7Du3EKtFUvon=-6-24QLs5m=Q@mail.gmail.com>
Message-ID: <1678477.gc6ClU3kc1@sos>

 Don't do it. Caller's phone carrier will drop the call if not answered in 1-2 
minutes.

On Saturday 11 August 2012 10:23:08 Vishal Kakkar wrote:
> Hi all,
> 
> I want to put the caller into fifo queue *without answering the call till
> agent attend it.*
> But fifo is answering the call while putting caller in queue.
> 
> Please help!!
> 
> Thanks,
> -Vishal.


From bdfoster at endigotech.com  Sat Aug 11 15:41:25 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 11 Aug 2012 07:41:25 -0400
Subject: [Freeswitch-users] Smack a Hacker context?
In-Reply-To: <CAKDfjgdVY4Y6GK8qdAAwfiFgSxHt-dnex_p+JTZo49bqNExi4Q@mail.gmail.com>
References: <211601cd7445$26907820$73b16860$@bizfocused.com>
	<CAKDfjgdVY4Y6GK8qdAAwfiFgSxHt-dnex_p+JTZo49bqNExi4Q@mail.gmail.com>
Message-ID: <CAKV-jFzf1PjpGXSNX4BCeQrdjUnynPKrhYvYu-4jvgGFHBjHEg@mail.gmail.com>

My servers are constantly getting hit by bots/wankers. It's not that big of
a deal if you know how to take care of it (fail2ban works wonders).
Obscurity is not really security. I've stopped sanitizing my logs for the
server ip's. I just don't care anymore.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 8, 2012 10:56 AM, "Kristian Kielhofner" <kris at kriskinc.com> wrote:

> In 99% of these cases your machine is being hit by a bot (sipvicious,
> etc).  Playing media is only going to waste your bandwidth and CPU
> resources (chances are they're using a rooted machine).  No human will
> ever hear your special message.
>
> Your "adult option" is the best bet.  Drop them, log them, and move on.
>
> On Mon, Aug 6, 2012 at 10:34 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> > HI Everyone,
> >
> >
> >
> > I didn?t bother to obliterate my servers real identity from my last
> pastebin
> > post, shame on me.  Of course within 24 hours some a$$hole is trying the
> > default login ids for Freeswitch.  I am not quite a noob here, but this
> > would take me some time to do and I know some of you guys could crank
> this
> > out in minutes.  Of course you all probably have better ideas too.
> >
> >
> >
> > I want to allow the users in the sample Freeswitch config to login ? but
> to
> > special context.  In the ?special? context, everything they dial plays a
> > recording like ?you?re an ass hangup and try again? or perhaps and ear
> > piercing LOUD tone.  Even better would be to dial them every few minutes
> > with the same recording or sound.  Best of all, if we can detect that
> they
> > are on a CISCO 5xx phone, I have a special config file to send them!
> >
> >
> >
> > Anyone got better ideas?  I would love to hear them.
> >
> >
> >
> > I suppose the adult choice would be to gather their external and
> internal ip
> > addresses and report them to their ISP, but that won?t achieve much.
>  Maybe
> > we could build a blacklist and watch for them to connect to the
> Freeswitch
> > conf call.  Then we could all tell them how much we enjoy their efforts.
> >
> >
> >
> > Sean
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vishal.kakkar at gmail.com  Sat Aug 11 15:48:30 2012
From: vishal.kakkar at gmail.com (Vishal Kakkar)
Date: Sat, 11 Aug 2012 17:18:30 +0530
Subject: [Freeswitch-users] mod_fifo
In-Reply-To: <1678477.gc6ClU3kc1@sos>
References: <CAAxwcquspdkRcNRfG9LA7kbie7Du3EKtFUvon=-6-24QLs5m=Q@mail.gmail.com>
	<1678477.gc6ClU3kc1@sos>
Message-ID: <CAAxwcqtO_q0k39qqVvxdJb3f7uQevn5=M3Q49ac+bom49VQ6Zg@mail.gmail.com>

<condition field="${destination_number}" expression="^8032342123$"
break="on-true">
<action application="ring_ready"/>
 <action application="sleep" data="3000"/>
 <action application="set" data="fifo_orbit_exten=_continue_:60"/>
* <action application="fifo" data="Testing in"/>*
</condition>

The moment sleep expires customer is queued and the call is answered(Dont
know why) irrespective of agents availability or not....
When any of Agents pick up and both are bridged by fifo..

I want customer call not to be answered(it should remain ringing) till any
agent picks the phone..

Please help,
-Vishal.


On Sat, Aug 11, 2012 at 3:56 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

>  Don't do it. Caller's phone carrier will drop the call if not answered in
> 1-2
> minutes.
>
> On Saturday 11 August 2012 10:23:08 Vishal Kakkar wrote:
> > Hi all,
> >
> > I want to put the caller into fifo queue *without answering the call till
> > agent attend it.*
> > But fifo is answering the call while putting caller in queue.
> >
> > Please help!!
> >
> > Thanks,
> > -Vishal.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From errotan at elder.hu  Sat Aug 11 16:52:50 2012
From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=)
Date: Sat, 11 Aug 2012 14:52:50 +0200
Subject: [Freeswitch-users] mod_fifo
In-Reply-To: <CAAxwcqtO_q0k39qqVvxdJb3f7uQevn5=M3Q49ac+bom49VQ6Zg@mail.gmail.com>
References: <CAAxwcquspdkRcNRfG9LA7kbie7Du3EKtFUvon=-6-24QLs5m=Q@mail.gmail.com>
	<1678477.gc6ClU3kc1@sos>
	<CAAxwcqtO_q0k39qqVvxdJb3f7uQevn5=M3Q49ac+bom49VQ6Zg@mail.gmail.com>
Message-ID: <502655A2.4050806@elder.hu>


As said by Sergey Okhapkin all telephone service providers terminate the 
call after
a few minutes of ringing if not answered so answering call is advised. 
Looking at
the code of mod_fifo the call is always answered so if you want this kind a
functionality open a feature request in jira ( 
http://jira.freeswitch.org/ ).


2012-08-11 13:48 keltez?ssel, Vishal Kakkar ?rta:
> <condition field="${destination_number}" expression="^8032342123$" 
> break="on-true">
> <action application="ring_ready"/>
>  <action application="sleep" data="3000"/>
>  <action application="set" data="fifo_orbit_exten=_continue_:60"/>
> * <action application="fifo" data="Testing in"/>*
> </condition>
>
> The moment sleep expires customer is queued and the call is 
> answered(Dont know why) irrespective of agents availability or not....
> When any of Agents pick up and both are bridged by fifo..
>
> I want customer call not to be answered(it should remain ringing) till 
> any agent picks the phone..
>
> Please help,
> -Vishal.
>
>
> On Sat, Aug 11, 2012 at 3:56 PM, Sergey Okhapkin 
> <sos at sokhapkin.dyndns.org <mailto:sos at sokhapkin.dyndns.org>> wrote:
>
>      Don't do it. Caller's phone carrier will drop the call if not
>     answered in 1-2
>     minutes.
>
>     On Saturday 11 August 2012 10:23:08 Vishal Kakkar wrote:
>     > Hi all,
>     >
>     > I want to put the caller into fifo queue *without answering the
>     call till
>     > agent attend it.*
>     > But fifo is answering the call while putting caller in queue.
>     >
>     > Please help!!
>     >
>     > Thanks,
>     > -Vishal.
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     FreeSWITCH-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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From admin at blindi.net  Sat Aug 11 18:16:52 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sat, 11 Aug 2012 16:16:52 +0200 (CEST)
Subject: [Freeswitch-users] dtmf delay abount more than 70 digits ivr
	transfers
In-Reply-To: <502655A2.4050806@elder.hu>
References: <CAAxwcquspdkRcNRfG9LA7kbie7Du3EKtFUvon=-6-24QLs5m=Q@mail.gmail.com>
	<1678477.gc6ClU3kc1@sos>
	<CAAxwcqtO_q0k39qqVvxdJb3f7uQevn5=M3Q49ac+bom49VQ6Zg@mail.gmail.com>
	<502655A2.4050806@elder.hu>
Message-ID: <Pine.LNX.4.64.1208111556160.5854@1a-8140.antagus.de>

Hi guys,
i have setup a voicechat.
the navigation is done by dtmf.
I transfer from ivr to ivr and back.
I have 2 transfers per one digit: example: to iv2 and ivr1 and so on.
After abbount 30 digits (60 transfers) the dtmf reaction 
is very delayd from fs, abbout 3 4 seconds.
80 dtmf tones (abbount 160 transfers) the delay is 10 or more seconds.

I don.t understand:
the delay is very high, then i jumpt in to a conference the delay will be 
reset.
I put the next digits the delay is higher and higher and so on.
The existing conference reset the delay at everytime.

I think a buffer problem?
Is this a bug in fs?
How  can i minimize the dtmf delay please?

thanks.


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From krice at freeswitch.org  Sat Aug 11 19:23:11 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 11 Aug 2012 10:23:11 -0500
Subject: [Freeswitch-users] Is authorizing presence subscriptions
	possible?
In-Reply-To: <52148.193.34.0.63.1344671661.squirrel@mail.dgtronik.com.pl>
References: <5024F755.8080003@dgt.com.pl>
	<D869AABB-A75F-4198-8638-B7E06287C146@freeswitch.org>
	<52148.193.34.0.63.1344671661.squirrel@mail.dgtronik.com.pl>
Message-ID: <1FA7AE64-E5CB-48FA-AAD5-F5118BD97C53@freeswitch.org>

i dont think that is possible as a per user scenario, but it may be possible via a per sofile profile... outside of that i would look at sponsoring one of the devs to add such a feature

Ken
Sent from my iPad

On Aug 11, 2012, at 2:54 AM, "Jakub Nadolski" <jakub.nadolski at dgt.com.pl> wrote:

> Yes, this is what I would like to do.
> 
> Kuba
> 
>> 
>> as in on a per user basis? ie user x can subscribe but user y cannot?
>> Ken
>> Sent from my iPad
>> 
>> On Aug 10, 2012, at 6:58 AM, Jakub Nadolski <jakub.nadolski at dgt.com.pl>
>> wrote:
>> 
>>> Hello everyone,
>>> 
>>> I have a question about presence in mod sofia. I use Aastra phones and
>>> BLF function works fine. But now I need to authorize subcriptions. So I
>>> need configuration in which not all users can subscribe and receive
>>> state of the extension in NOTIFY messages. Is that possible and if so
>>> how can I do that?
>>> 
>>> Thanks in advance,
>>> Kuba
>>> 
>>> 
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> Join Us At ClueCon - Aug 7-9, 2012
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Sat Aug 11 23:12:42 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 11 Aug 2012 14:12:42 -0500
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAKV-jFyJEGW72UwZ8AGkKPL9uvoab3fOUb7y5_wTx6_suPT0JQ@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F15141094@cantor.std.visionutv.se>
	<50236299.4070403@softnet.si>
	<CAKV-jFyJEGW72UwZ8AGkKPL9uvoab3fOUb7y5_wTx6_suPT0JQ@mail.gmail.com>
Message-ID: <CAKbxfG8Wavpt7COjPxm9BGA7LV-8_NZBS7koakMNTiTUBE-4=A@mail.gmail.com>

Based on the inexplicable issues in cent6 working on debian packages might
be very fruitful.  Talk to krice at freeswitch.org
On Aug 10, 2012 8:32 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> I need to update the wiki, but the install script for Debian is at
> github.com/bdfoster/fs-debian-installer. It still gets the latest code on
> master. If you guys want to give me some feedback I can incorporate the
> proper changes. I stopped working on it because most are using centos and I
> am not going to maintain something no one uses. I have the time, I have the
> energy, but I need to know what people would like to see. Our internal
> script is based on this one but I cannot release those changes to the world.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Aug 9, 2012 3:12 AM, "Miha" <miha at softnet.si> wrote:
>
>>  On 8/9/2012 8:36 AM, Peter Olsson wrote:
>>
>>  If you change to branch ?v1.2.stable?, the version will be 1.2.0.****
>>
>> ** **
>>
>> The core devs have not yet sent out any official instructions, but I
>> guess that if you want to continue to follow the main development (with
>> sometimes unstable code), just continue to do as you do today. If you want
>> to stick to the 1.2 version, and only get bugfixes etc. for that version
>> (but no new features), change to the branch v1.2.stable.****
>>
>> ** **
>>
>> I think there will be more official instructions soon, but this is that
>> way I think it?s meant to work.****
>>
>> ** **
>>
>> /Peter****
>>
>> ** **
>>
>> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [
>> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
>> *F?r *Miha
>> *Skickat:* den 9 augusti 2012 08:15
>> *Till:* FreeSWITCH Users Help
>> *?mne:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2****
>>
>> ** **
>>
>> Hi,
>>
>> so my current version is 1.2 rc2. After make current I will have 1.2? I
>> try it on test server but still FreeSWITCH Version
>> 1.2.0-rc2+git~20120809T022906Z~3389f32363 (1.2.0-rc2; git at commit
>> 3389f32363 on Thu, 09 Aug 2012 02:29:06 Z)
>>
>>
>> Thanks!
>>
>> Miha
>>
>> On 8/8/2012 5:02 PM, Ken Rice wrote:****
>>
>> the Primary repo is git.freeswitch.org/freeswitch.git this is the master
>> repo, however, the github.com/freeswitch/freeswitch is an unofficial
>> mirror...  updates to primary git repo should be pushed including tags and
>> branches but we do not support those that wish to use other than official
>> repos ****
>>
>> ** **
>>
>> ** **
>>
>> K****
>>
>> ** **
>>
>> On Wed, Aug 8, 2012 at 8:39 AM, Brian Foster <bdfoster at endigotech.com>
>> wrote:****
>>
>> Ah, looks like they publish to github as well. Afaik the primary repo is
>> git.freeswitch.org, and that is where I get everything. ****
>>
>> Brian Foster
>> Endigo Computer LLC****
>>
>> Sent from a mobile device.****
>>
>> On Aug 8, 2012 8:41 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:****
>>
>> What is this then? https://github.com/freeswitch/freeswitch ****
>>
>>
>> Regards,
>> Ben Langfeld
>>
>> ****
>>
>> On 8 August 2012 12:53, Brian Foster <bdfoster at endigotech.com> wrote:****
>>
>> FreeSWITCH isn't published on GitHub.****
>>
>> Brian Foster
>> Endigo Computer LLC****
>>
>> Sent from a mobile device.****
>>
>> On Aug 8, 2012 3:14 AM, "Stanislav Sinyagin" <ssinyagin at yahoo.com> wrote:
>> ****
>>
>> Ken,****
>>
>> care to push tags to Github as well?****
>>
>> ** **
>>   ------------------------------
>>
>> *From:* Ken Rice <krice at freeswitch.org>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Tuesday, August 7, 2012 8:31 PM
>> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2****
>>
>> ** **
>>
>> yes its there... git pull --tags****
>>
>> ** **
>>
>> On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net> wrote:***
>> *
>>
>> I also checked tags, but didn't see it. Are you sure your branch is
>> tracked upstream?****
>>
>>  ****
>>
>> --Dave****
>>
>> ** **
>>  ------------------------------
>>
>> *From:* Ken Rice [mailto:krice at freeswitch.org]
>> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org
>> ]****
>>
>> *Sent:* Tue, 07 Aug 2012 11:15:50 -0700
>> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2 ****
>>
>>
>>
>> its Tagged, right now, we will be branching it shortly for tracking
>> patches. Please note we had a couple of good users point out some smallish
>> bugs so we're int he process of fixing those! so guys if you find anything
>> wrong please open a jira, if you are at cluecon, find me and let me know
>> the bug number ****
>>
>> ** **
>>
>> K****
>>
>> On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net> wrote:***
>> *
>>
>> Where is the branch so I can checkout local?****
>>
>>  ****
>>
>> D:\fsnew>git branch -a
>> * master
>>   remotes/origin/FS-3432
>>   remotes/origin/FS-4062
>>   remotes/origin/HEAD -> origin/master
>>   remotes/origin/dingaling_video
>>   remotes/origin/master
>>   remotes/origin/smgfs
>>   remotes/origin/stable-test/freeswitch-1.2
>>   remotes/origin/swk/fs_test
>>   remotes/test/master****
>>
>> --Dave****
>>
>> ** **
>>  ------------------------------
>>
>> *From:* Ken Rice [mailto:krice at freeswitch.org]
>> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org],
>> freeswitch-dev at lists.freeswitch.org
>> *Sent:* Tue, 07 Aug 2012 09:03:34 -0700 ****
>>
>>
>> *Subject:* [Freeswitch-users] Announcing FreeSWITCH 1.2****
>>
>> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0! ** **
>>
>> ** **
>>
>> Get your copy today at
>> http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 ! ****
>>
>> ** **
>>
>> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
>> continue to fix bugs and security issues. giving you a stable platform for
>> at least one year.****
>>
>> ** **
>>
>> Grab it today!****
>>
>> ** **
>>
>> The FreeSWITCH Team ****
>>
>>   ****
>>
>>  ****
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com/
>>
>> 
>> /
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org/
>> http://wiki.freeswitch.org/
>> http://www.cluecon.com/
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>> ** **
>>
>>   ****
>>
>>  ****
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>> ** **
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> ****
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>> ** **
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org****
>>
>> ** **
>>
>>
>>
>>
>> ****
>>
>> _________________________________________________________________________****
>>
>> Professional FreeSWITCH Consulting Services:****
>>
>> consulting at freeswitch.org****
>>
>> http://www.freeswitchsolutions.com****
>>
>> ** **
>>
>> ****
>>
>> ****
>>
>> ** **
>>
>> Official FreeSWITCH Sites****
>>
>> http://www.freeswitch.org****
>>
>> http://wiki.freeswitch.org****
>>
>> http://www.cluecon.com****
>>
>> ** **
>>
>> Join Us At ClueCon - Aug 7-9, 2012****
>>
>> ** **
>>
>> FreeSWITCH-users mailing list****
>>
>> FreeSWITCH-users at 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:502354a432765721010175! ****
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> 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
>>
>>  @Peter,
>>
>> thanks for all your explanation.
>>
>> Br,
>> Miha
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Aug 11 23:19:44 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 11 Aug 2012 14:19:44 -0500
Subject: [Freeswitch-users] mod_fifo
In-Reply-To: <502655A2.4050806@elder.hu>
References: <CAAxwcquspdkRcNRfG9LA7kbie7Du3EKtFUvon=-6-24QLs5m=Q@mail.gmail.com>
	<1678477.gc6ClU3kc1@sos>
	<CAAxwcqtO_q0k39qqVvxdJb3f7uQevn5=M3Q49ac+bom49VQ6Zg@mail.gmail.com>
	<502655A2.4050806@elder.hu>
Message-ID: <CAKbxfG8CK0dEEY2wspT8fpoW2CaStVgS2iw6u-v9uvgsBZYANA@mail.gmail.com>

I will not add this feature so don't bother.

This is a form of toll fraud to host services and avoid answering calls.
On Aug 11, 2012 7:54 AM, "Pusk?s Zsolt" <errotan at elder.hu> wrote:

>
> As said by Sergey Okhapkin all telephone service providers terminate the
> call after
> a few minutes of ringing if not answered so answering call is advised.
> Looking at
> the code of mod_fifo the call is always answered so if you want this kind a
> functionality open a feature request in jira ( http://jira.freeswitch.org/).
>
>
> 2012-08-11 13:48 keltez?ssel, Vishal Kakkar ?rta:
>
> <condition field="${destination_number}" expression="^8032342123$"
> break="on-true">
> <action application="ring_ready"/>
>  <action application="sleep" data="3000"/>
>  <action application="set" data="fifo_orbit_exten=_continue_:60"/>
> * <action application="fifo" data="Testing in"/>*
> </condition>
>
> The moment sleep expires customer is queued and the call is answered(Dont
> know why) irrespective of agents availability or not....
> When any of Agents pick up and both are bridged by fifo..
>
> I want customer call not to be answered(it should remain ringing) till any
> agent picks the phone..
>
> Please help,
> -Vishal.
>
>
>  On Sat, Aug 11, 2012 at 3:56 PM, Sergey Okhapkin <
> sos at sokhapkin.dyndns.org> wrote:
>
>>  Don't do it. Caller's phone carrier will drop the call if not answered
>> in 1-2
>> minutes.
>>
>> On Saturday 11 August 2012 10:23:08 Vishal Kakkar wrote:
>> > Hi all,
>> >
>>  > I want to put the caller into fifo queue *without answering the call
>> till
>> > agent attend it.*
>> > But fifo is answering the call while putting caller in queue.
>> >
>> > Please help!!
>> >
>> > Thanks,
>> > -Vishal.
>>
>>  _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lynn.nielson at greenseedtechnologies.com  Sun Aug 12 00:10:45 2012
From: lynn.nielson at greenseedtechnologies.com (Lynn Nielson)
Date: Sat, 11 Aug 2012 14:10:45 -0600
Subject: [Freeswitch-users] uuid_media off appears to cancel
	park_after_bridge
Message-ID: <5026BC45.6030208@greenseedtechnologies.com>

We are building a dialer interface to our client app that calls our 
customers and then allows those customers to connect and autodial to 
their clients.  We are using these commands to do this:

originate 
{park_after_bridge=true}sofia/gateway/vitelity-outbound/xxxxxxxxxx &park
    +OK <uuid_1>
originate sofia/gateway/vitelity-outbound/yyyyyyyyyy &park
    +OK <uuid_2>
uuid_bridge <uuid_1> <uuid_2>
uuid_media off <uuid_2>

If the second call (leg-b) hangs up, then both calls are dropped and the 
park_after_bridge=true appears to be ignored.  If we leave off the last 
command of uuid_media off <uuid>, then the call behaves as expected on 
hang-up leaving the first call (leg-a) open. My problem is the we have 
no audio on the call when we do not add the uuid_media off command.

This only happens when bridging two external calls.  If one of the legs 
is an internal extension, then we don't need the uuid_media off and 
audio works fine.

Is there another way of doing this or a way for uuid_media off to honor 
the park_after_bridge condition?

btw: we have also tried this using Flowroute as the gateway as well.

Thanks

Lynn


From krice at freeswitch.org  Sun Aug 12 00:23:47 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 11 Aug 2012 15:23:47 -0500
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAKbxfG8Wavpt7COjPxm9BGA7LV-8_NZBS7koakMNTiTUBE-4=A@mail.gmail.com>
Message-ID: <CC4C2983.3AFAB%krice@freeswitch.org>

I?ll catch up with you on IRC Brian..


On 8/11/12 2:12 PM, "Anthony Minessale" <anthony.minessale at gmail.com> wrote:

> Based on the inexplicable issues in cent6 working on debian packages might be
> very fruitful.? Talk to krice at freeswitch.org
> 
> On Aug 10, 2012 8:32 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>> 
>> I need to update the wiki, but the install script for Debian is at
>> github.com/bdfoster/fs-debian-installer
>> <http://github.com/bdfoster/fs-debian-installer> . It still gets the latest
>> code on master. If you guys want to give me some feedback I can incorporate
>> the proper changes. I stopped working on it because most are using centos and
>> I am not going to maintain something no one uses. I have the time, I have the
>> energy, but I need to know what people would like to see. Our internal script
>> is based on this one but I cannot release those changes to the world.
>> 
>> Brian Foster
>> Endigo Computer LLC
>> 
>> Sent from a mobile device.
>> 
>> On Aug 9, 2012 3:12 AM, "Miha" <miha at softnet.si> wrote:
>>>     
>>>  
>>> On 8/9/2012 8:36 AM, Peter Olsson wrote:
>>>  
>>>  
>>>>      
>>>>  
>>>> 
>>>> If you change to branch ?v1.2.stable?, the version will be 1.2.0.
>>>>  
>>>> ?
>>>>  
>>>> The core devs have not yet sent out any official instructions, but I guess
>>>> that if you want to continue to follow the main development (with sometimes
>>>> unstable code), just continue to do as you do today. If you want to stick
>>>> to the 1.2 version, and only get bugfixes etc. for that version (but no new
>>>> features), change to the branch v1.2.stable.
>>>>  
>>>> ?
>>>>  
>>>> I think there will be more official instructions soon, but this is that way
>>>> I think it?s meant to work.
>>>>  
>>>> ?
>>>>  
>>>> /Peter
>>>>  
>>>> ?
>>>>  
>>>>  
>>>>  
>>>> 
>>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
>>>>  Skickat: den 9 augusti 2012 08:15
>>>>  Till: FreeSWITCH Users Help
>>>>  ?mne: Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>>>>  
>>>>  
>>>>  
>>>> ?
>>>>  
>>>>  
>>>> 
>>>> Hi,
>>>>  
>>>>  so my current version is 1.2 rc2. After make current I will have 1.2? I
>>>> try it on test server but still FreeSWITCH Version
>>>> 1.2.0-rc2+git~20120809T022906Z~3389f32363 (1.2.0-rc2; git at commit
>>>> 3389f32363 on Thu, 09 Aug 2012 02:29:06 Z)
>>>>  
>>>>  
>>>>  Thanks!
>>>>  
>>>>  Miha
>>>>  
>>>>  On 8/8/2012 5:02 PM, Ken Rice wrote:
>>>>  
>>>>  
>>>>>  
>>>>> the Primary repo is  git.freeswitch.org/freeswitch.git
>>>>> <http://git.freeswitch.org/freeswitch.git>  this is the master repo,
>>>>> however, the  github.com/freeswitch/freeswitch
>>>>> <http://github.com/freeswitch/freeswitch>  is an unofficial mirror...
>>>>> ?updates to primary git repo should be pushed including tags and branches
>>>>> but we do not support those that wish to use other than official repos
>>>>>  
>>>>>  
>>>>> 
>>>>> ?
>>>>>  
>>>>>  
>>>>>  
>>>>> 
>>>>> ?
>>>>>  
>>>>>  
>>>>>  
>>>>> 
>>>>> K
>>>>>  
>>>>>  
>>>>>  
>>>>> 
>>>>> ?
>>>>>  
>>>>>  
>>>>> 
>>>>> On Wed, Aug 8, 2012 at 8:39 AM, Brian Foster <bdfoster at endigotech.com>
>>>>> wrote:
>>>>>  
>>>>> Ah, looks like they publish to github as well. Afaik the primary repo is
>>>>> git.freeswitch.org <http://git.freeswitch.org> , and that is where I get
>>>>> everything. 
>>>>>  
>>>>>  
>>>>> 
>>>>> Brian Foster
>>>>>  Endigo Computer LLC
>>>>>  
>>>>> 
>>>>> Sent from a mobile device.
>>>>>  
>>>>>  
>>>>>  
>>>>>  
>>>>>  
>>>>> 
>>>>> On Aug 8, 2012 8:41 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>>>>>  
>>>>> What is this then??https://github.com/freeswitch/freeswitch
>>>>>  
>>>>>  
>>>>> 
>>>>> 
>>>>>  Regards,
>>>>>  Ben Langfeld
>>>>>  
>>>>>  
>>>>>  
>>>>>  
>>>>> 
>>>>> On 8 August 2012 12:53, Brian Foster <bdfoster at endigotech.com> wrote:
>>>>>  
>>>>> FreeSWITCH isn't published on GitHub.
>>>>>  
>>>>> 
>>>>> Brian Foster
>>>>>  Endigo Computer LLC
>>>>>  
>>>>> 
>>>>> Sent from a mobile device.
>>>>>  
>>>>>  
>>>>>  
>>>>>  
>>>>> 
>>>>> On Aug 8, 2012 3:14 AM, "Stanislav Sinyagin" <ssinyagin at yahoo.com> wrote:
>>>>>  
>>>>>  
>>>>>  
>>>>>  
>>>>> 
>>>>> Ken,
>>>>>  
>>>>>  
>>>>>  
>>>>> 
>>>>> care to push tags to Github as well?
>>>>>  
>>>>>  
>>>>>  
>>>>>>  
>>>>>> 
>>>>>> ?
>>>>>>  
>>>>>>  
>>>>>>  
>>>>>>  
>>>>>>  
>>>>>> 
>>>>>>  
>>>>>>  
>>>>>> 
>>>>>> From: Ken Rice <krice at freeswitch.org>
>>>>>>  To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>>>  Sent: Tuesday, August 7, 2012 8:31 PM
>>>>>>  Subject: Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>>>>>>  
>>>>>>  
>>>>>> ?
>>>>>>  
>>>>>>  
>>>>>> 
>>>>>> yes its there... git pull --tags
>>>>>>  
>>>>>>  
>>>>>> 
>>>>>> ?
>>>>>>  
>>>>>>  
>>>>>> 
>>>>>> On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>>>>>>  
>>>>>>  
>>>>>>  
>>>>>> 
>>>>>> I also checked tags, but didn't see it. Are you sure your branch is
>>>>>> tracked upstream?
>>>>>>  
>>>>>>  
>>>>>>  
>>>>>> 
>>>>>> ?
>>>>>>  
>>>>>>  
>>>>>>  
>>>>>> 
>>>>>> --Dave
>>>>>>  
>>>>>>  
>>>>>>>  
>>>>>>> ?
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> From: Ken Rice [mailto:krice at freeswitch.org]
>>>>>>>  To: FreeSWITCH Users Help
>>>>>>> [mailto:freeswitch-users at lists.freeswitch.org]
>>>>>>>  
>>>>>>>  
>>>>>>> Sent: Tue, 07 Aug 2012 11:15:50 -0700
>>>>>>>  Subject: Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> 
>>>>>>>  
>>>>>>>  its Tagged, right now, we will be branching it shortly for tracking
>>>>>>> patches. Please note we had a couple of good users point out some
>>>>>>> smallish bugs so we're int he process of fixing those! so guys if you
>>>>>>> find anything wrong please open a jira, if you are at cluecon, find me
>>>>>>> and let me know the bug number
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> ?
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> K
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> Where is the branch so I can checkout local?
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> ?
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> D:\fsnew>git branch -a
>>>>>>>  * master
>>>>>>>  ? remotes/origin/FS-3432
>>>>>>>  ? remotes/origin/FS-4062
>>>>>>>  ? remotes/origin/HEAD -> origin/master
>>>>>>>  ? remotes/origin/dingaling_video
>>>>>>>  ? remotes/origin/master
>>>>>>>  ? remotes/origin/smgfs
>>>>>>>  ? remotes/origin/stable-test/freeswitch-1.2
>>>>>>>  ? remotes/origin/swk/fs_test
>>>>>>>  ? remotes/test/master
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> --Dave
>>>>>>>  
>>>>>>>  
>>>>  
>>>> ?
>>>>  
>>>>  
>>>> 
>>>>  
>>>>  
>>>> 
>>>> From: Ken Rice [mailto:krice at freeswitch.org]
>>>>  To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org],
>>>> freeswitch-dev at lists.freeswitch.org
>>>>  Sent: Tue, 07 Aug 2012 09:03:34 -0700
>>>>  
>>>>  
>>>> 
>>>> 
>>>>  Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
>>>>  
>>>>  
>>>>  
>>>>  
>>>> 
>>>> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!
>>>>  
>>>>  
>>>> 
>>>> ?
>>>>  
>>>>  
>>>>  
>>>> 
>>>> Get your copy today at?http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2
>>>> !?
>>>>  
>>>>  
>>>>  
>>>> 
>>>> ?
>>>>  
>>>>  
>>>>  
>>>> 
>>>> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
>>>> continue to fix bugs and security issues. giving you a stable platform for
>>>> at least one year.
>>>>  
>>>>  
>>>>  
>>>> 
>>>> ?
>>>>  
>>>>  
>>>>  
>>>> 
>>>> Grab it today!
>>>>  
>>>>  
>>>>  
>>>> 
>>>> ?
>>>>  
>>>>  
>>>>  
>>>> 
>>>> The FreeSWITCH Team?
>>>>  
>>>>  
>>>>  
>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> ?
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> ?
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> 
________________________________________________________________________>>>>>>>
_
>>>>>>>  Professional FreeSWITCH Consulting Services:
>>>>>>>  consulting at freeswitch.org
>>>>>>>  http://www.freeswitchsolutions.com/
>>>>>>>  
>>>>>>>  
>>>>>>>  /
>>>>>>>  
>>>>>>>  Official FreeSWITCH Sites
>>>>>>>  http://www.freeswitch.org/
>>>>>>>  http://wiki.freeswitch.org/
>>>>>>>  http://www.cluecon.com/
>>>>>>>  
>>>>>>>  Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>  
>>>>>>>  FreeSWITCH-users mailing list
>>>>>>>  FreeSWITCH-users at 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/>
>>>>>>>  
>>>>>>>  
>>>>>>> ?
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> ?
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> ?
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> 
________________________________________________________________________>>>>>>>
_
>>>>>>>  Professional FreeSWITCH Consulting Services:
>>>>>>>  consulting at freeswitch.org
>>>>>>>  http://www.freeswitchsolutions.com
>>>>>>> <http://www.freeswitchsolutions.com/>
>>>>>>>  
>>>>>>>  
>>>>>>>   </>
>>>>>>>  
>>>>>>>  Official FreeSWITCH Sites
>>>>>>>  http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>>>  http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>>>>>>>  http://www.cluecon.com <http://www.cluecon.com/>
>>>>>>>  
>>>>>>>  Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>  
>>>>>>>  FreeSWITCH-users mailing list
>>>>>>>  FreeSWITCH-users at 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/>
>>>>>>>  
>>>>>>>  
>>>>>>> ?
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> 
________________________________________________________________________>>>>>>>
_
>>>>>>>  Professional FreeSWITCH Consulting Services:
>>>>>>>  consulting at freeswitch.org
>>>>>>>  http://www.freeswitchsolutions.com
>>>>>>> <http://www.freeswitchsolutions.com/>
>>>>>>>  
>>>>>>>  
>>>>>>>   </>
>>>>>>>  
>>>>>>>  Official FreeSWITCH Sites
>>>>>>>  http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>>>  http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>>>>>>>  http://www.cluecon.com <http://www.cluecon.com/>
>>>>>>>  
>>>>>>>  Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>  
>>>>>>>  FreeSWITCH-users mailing list
>>>>>>>  FreeSWITCH-users at 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/>
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> 
________________________________________________________________________>>>>>>>
_
>>>>>>>  Professional FreeSWITCH Consulting Services:
>>>>>>>  consulting at freeswitch.org
>>>>>>>  http://www.freeswitchsolutions.com
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  Official FreeSWITCH Sites
>>>>>>>  http://www.freeswitch.org
>>>>>>>  http://wiki.freeswitch.org
>>>>>>>  http://www.cluecon.com
>>>>>>>  
>>>>>>>  Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>  
>>>>>>>  FreeSWITCH-users mailing list
>>>>>>>  FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>  
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  http://www.freeswitch.org
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> 
________________________________________________________________________>>>>>>>
_
>>>>>>>  Professional FreeSWITCH Consulting Services:
>>>>>>>  consulting at freeswitch.org
>>>>>>>  http://www.freeswitchsolutions.com
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  Official FreeSWITCH Sites
>>>>>>>  http://www.freeswitch.org
>>>>>>>  http://wiki.freeswitch.org
>>>>>>>  http://www.cluecon.com
>>>>>>>  
>>>>>>>  Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>  
>>>>>>>  FreeSWITCH-users mailing list
>>>>>>>  FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>  
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  http://www.freeswitch.org
>>>>>>>  
>>>>>>>  
>>>>>>> ?
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> 
________________________________________________________________________>>>>>>>
_
>>>>>>>  Professional FreeSWITCH Consulting Services:
>>>>>>>  consulting at freeswitch.org
>>>>>>>  http://www.freeswitchsolutions.com
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  Official FreeSWITCH Sites
>>>>>>>  http://www.freeswitch.org
>>>>>>>  http://wiki.freeswitch.org
>>>>>>>  http://www.cluecon.com
>>>>>>>  
>>>>>>>  Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>  
>>>>>>>  FreeSWITCH-users mailing list
>>>>>>>  FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>  
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  http://www.freeswitch.org
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>> 
________________________________________________________________________>>>>>>>
_
>>>>>>>  Professional FreeSWITCH Consulting Services:
>>>>>>>  consulting at freeswitch.org
>>>>>>>  http://www.freeswitchsolutions.com
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  Official FreeSWITCH Sites
>>>>>>>  http://www.freeswitch.org
>>>>>>>  http://wiki.freeswitch.org
>>>>>>>  http://www.cluecon.com
>>>>>>>  
>>>>>>>  Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>  
>>>>>>>  FreeSWITCH-users mailing list
>>>>>>>  FreeSWITCH-users at lists.freeswitch.org
>>>>>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>  
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>  http://www.freeswitch.org
>>>>>>>  
>>>>>>>  
>>>>>>> ?
>>>>>>>  
>>>>>>>  
>>>>>>> 
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>>  
>>>>>>> 
________________________________________________________________________>>>>>>>
_
>>>>>>>  
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>  
>>>>>>> consulting at freeswitch.org
>>>>>>>  
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>  
>>>>>>> ?
>>>>>>>  
>>>>>>> 
>>>>>>>  
>>>>>>> 
>>>>>>>  
>>>>>>> ?
>>>>>>>  
>>>>>>> Official FreeSWITCH Sites
>>>>>>>  
>>>>>>> http://www.freeswitch.org
>>>>>>>  
>>>>>>> http://wiki.freeswitch.org
>>>>>>>  
>>>>>>> http://www.cluecon.com
>>>>>>>  
>>>>>>> ?
>>>>>>>  
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>>  
>>>>>>> ?
>>>>>>>  
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>  
>>>>>>> FreeSWITCH-users at 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:502354a432765721010175!
>>>>>>>  
>>>>>>>  
>>>>>>>   
>>>>>>>  
>>>>>>> 
________________________________________________________________________>>>>>>>
_
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>> 
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>  
>>>>>>>  @Peter,
>>>>>>>  
>>>>>>>  thanks for all your explanation.
>>>>>>>  
>>>>>>>  Br,
>>>>>>>  Miha
>>>>>>>  
>>>>>>> 
>>>>>>> 
________________________________________________________________________>>>>>>>
_
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>> 
>>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.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.king at quentustech.com  Sun Aug 12 01:00:08 2012
From: william.king at quentustech.com (William King)
Date: Sat, 11 Aug 2012 16:00:08 -0500
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAKV-jFyJEGW72UwZ8AGkKPL9uvoab3fOUb7y5_wTx6_suPT0JQ@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F15141094@cantor.std.visionutv.se>
	<50236299.4070403@softnet.si>
	<CAKV-jFyJEGW72UwZ8AGkKPL9uvoab3fOUb7y5_wTx6_suPT0JQ@mail.gmail.com>
Message-ID: <5026C7D8.80008@quentustech.com>

There is work that is nearly completed to get debian repos setup for the 
stable releases and the nightly releases. Once that is wrapped up the 
wiki can be updated to reflect the new repo locations.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 08/10/2012 08:31 PM, Brian Foster wrote:
>
> I need to update the wiki, but the install script for Debian is at 
> github.com/bdfoster/fs-debian-installer 
> <http://github.com/bdfoster/fs-debian-installer>. It still gets the 
> latest code on master. If you guys want to give me some feedback I can 
> incorporate the proper changes. I stopped working on it because most 
> are using centos and I am not going to maintain something no one uses. 
> I have the time, I have the energy, but I need to know what people 
> would like to see. Our internal script is based on this one but I 
> cannot release those changes to the world.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
>
> On Aug 9, 2012 3:12 AM, "Miha" <miha at softnet.si 
> <mailto:miha at softnet.si>> wrote:
>
>     On 8/9/2012 8:36 AM, Peter Olsson wrote:
>>
>>     If you change to branch "v1.2.stable", the version will be 1.2.0.
>>
>>     The core devs have not yet sent out any official instructions,
>>     but I guess that if you want to continue to follow the main
>>     development (with sometimes unstable code), just continue to do
>>     as you do today. If you want to stick to the 1.2 version, and
>>     only get bugfixes etc. for that version (but no new features),
>>     change to the branch v1.2.stable.
>>
>>     I think there will be more official instructions soon, but this
>>     is that way I think it's meant to work.
>>
>>     /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] *F?r *Miha
>>     *Skickat:* den 9 augusti 2012 08:15
>>     *Till:* FreeSWITCH Users Help
>>     *?mne:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>>
>>     Hi,
>>
>>     so my current version is 1.2 rc2. After make current I will have
>>     1.2? I try it on test server but still FreeSWITCH Version
>>     1.2.0-rc2+git~20120809T022906Z~3389f32363 (1.2.0-rc2; git at
>>     commit 3389f32363 on Thu, 09 Aug 2012 02:29:06 Z)
>>
>>
>>     Thanks!
>>
>>     Miha
>>
>>     On 8/8/2012 5:02 PM, Ken Rice wrote:
>>
>>         the Primary repo is git.freeswitch.org/freeswitch.git
>>         <http://git.freeswitch.org/freeswitch.git> this is the master
>>         repo, however, the github.com/freeswitch/freeswitch
>>         <http://github.com/freeswitch/freeswitch> is an unofficial
>>         mirror...  updates to primary git repo should be pushed
>>         including tags and branches but we do not support those that
>>         wish to use other than official repos
>>
>>         K
>>
>>         On Wed, Aug 8, 2012 at 8:39 AM, Brian Foster
>>         <bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>> wrote:
>>
>>         Ah, looks like they publish to github as well. Afaik the
>>         primary repo is git.freeswitch.org
>>         <http://git.freeswitch.org>, and that is where I get everything.
>>
>>         Brian Foster
>>         Endigo Computer LLC
>>
>>         Sent from a mobile device.
>>
>>         On Aug 8, 2012 8:41 AM, "Ben Langfeld" <ben at langfeld.co.uk
>>         <mailto:ben at langfeld.co.uk>> wrote:
>>
>>         What is this then? https://github.com/freeswitch/freeswitch
>>
>>
>>         Regards,
>>         Ben Langfeld
>>
>>         On 8 August 2012 12:53, Brian Foster <bdfoster at endigotech.com
>>         <mailto:bdfoster at endigotech.com>> wrote:
>>
>>         FreeSWITCH isn't published on GitHub.
>>
>>         Brian Foster
>>         Endigo Computer LLC
>>
>>         Sent from a mobile device.
>>
>>         On Aug 8, 2012 3:14 AM, "Stanislav Sinyagin"
>>         <ssinyagin at yahoo.com <mailto:ssinyagin at yahoo.com>> wrote:
>>
>>         Ken,
>>
>>         care to push tags to Github as well?
>>
>>             ------------------------------------------------------------------------
>>
>>             *From:*Ken Rice <krice at freeswitch.org
>>             <mailto:krice at freeswitch.org>>
>>             *To:* FreeSWITCH Users Help
>>             <freeswitch-users at lists.freeswitch.org
>>             <mailto:freeswitch-users at lists.freeswitch.org>>
>>             *Sent:* Tuesday, August 7, 2012 8:31 PM
>>             *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>>
>>             yes its there... git pull --tags
>>
>>             On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel
>>             <drk at drkngs.net <mailto:drk at drkngs.net>> wrote:
>>
>>             I also checked tags, but didn't see it. Are you sure your
>>             branch is tracked upstream?
>>
>>             --Dave
>>
>>                 ------------------------------------------------------------------------
>>
>>                 *From:* Ken Rice [mailto:krice at freeswitch.org
>>                 <mailto:krice at freeswitch.org>]
>>                 *To:* FreeSWITCH Users Help
>>                 [mailto:freeswitch-users at lists.freeswitch.org
>>                 <mailto:freeswitch-users at lists.freeswitch.org>]
>>
>>                 *Sent:* Tue, 07 Aug 2012 11:15:50 -0700
>>                 *Subject:* Re: [Freeswitch-users] Announcing
>>                 FreeSWITCH 1.2
>>
>>
>>
>>                 its Tagged, right now, we will be branching it
>>                 shortly for tracking patches. Please note we had a
>>                 couple of good users point out some smallish bugs so
>>                 we're int he process of fixing those! so guys if you
>>                 find anything wrong please open a jira, if you are at
>>                 cluecon, find me and let me know the bug number
>>
>>                 K
>>
>>                 On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel
>>                 <drk at drkngs.net <mailto:drk at drkngs.net>> wrote:
>>
>>                 Where is the branch so I can checkout local?
>>
>>                 D:\fsnew>git branch -a
>>                 * master
>>                   remotes/origin/FS-3432
>>                   remotes/origin/FS-4062
>>                   remotes/origin/HEAD -> origin/master
>>                   remotes/origin/dingaling_video
>>                   remotes/origin/master
>>                   remotes/origin/smgfs
>>                   remotes/origin/stable-test/freeswitch-1.2
>>                   remotes/origin/swk/fs_test
>>                   remotes/test/master
>>
>>                 --Dave
>>
>>                     ------------------------------------------------------------------------
>>
>>                     *From:* Ken Rice [mailto:krice at freeswitch.org
>>                     <mailto:krice at freeswitch.org>]
>>                     *To:* FreeSWITCH Users Help
>>                     [mailto:freeswitch-users at lists.freeswitch.org
>>                     <mailto:freeswitch-users at lists.freeswitch.org>],
>>                     freeswitch-dev at lists.freeswitch.org
>>                     <mailto:freeswitch-dev at lists.freeswitch.org>
>>                     *Sent:* Tue, 07 Aug 2012 09:03:34 -0700
>>
>>
>>                     *Subject:* [Freeswitch-users] Announcing
>>                     FreeSWITCH 1.2
>>
>>                     The FreeSWITCH Team is Proud to announce
>>                     FreeSWITCH 1.2.0!
>>
>>                     Get your copy today at
>>                     http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2
>>                     !
>>
>>                     Going forward FreeSWITCH 1.2.x branch will be
>>                     feature stable, but we will continue to fix bugs
>>                     and security issues. giving you a stable platform
>>                     for at least one year.
>>
>>                     Grab it today!
>>
>>                     The FreeSWITCH Team
>>
>>
>>                 _________________________________________________________________________
>>                 Professional FreeSWITCH Consulting Services:
>>                 consulting at freeswitch.org
>>                 <mailto:consulting at freeswitch.org>
>>                 http://www.freeswitchsolutions.com/
>>
>>                 FreeSWITCH-powered IP PBX: The CudaTel Communication
>>                 Server
>>                 /
>>
>>                 Official FreeSWITCH Sites
>>                 http://www.freeswitch.org/
>>                 http://wiki.freeswitch.org/
>>                 http://www.cluecon.com/
>>
>>                 Join Us At ClueCon - Aug 7-9, 2012
>>
>>                 FreeSWITCH-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/>
>>
>>
>>             _________________________________________________________________________
>>             Professional FreeSWITCH Consulting Services:
>>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>             http://www.freeswitchsolutions.com
>>             <http://www.freeswitchsolutions.com/>
>>
>>             
>>              </>
>>
>>             Official FreeSWITCH Sites
>>             http://www.freeswitch.org <http://www.freeswitch.org/>
>>             http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>>             http://www.cluecon.com <http://www.cluecon.com/>
>>
>>             Join Us At ClueCon - Aug 7-9, 2012
>>
>>             FreeSWITCH-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/>
>>
>>
>>             _________________________________________________________________________
>>             Professional FreeSWITCH Consulting Services:
>>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>             http://www.freeswitchsolutions.com
>>             <http://www.freeswitchsolutions.com/>
>>
>>             
>>              </>
>>
>>             Official FreeSWITCH Sites
>>             http://www.freeswitch.org <http://www.freeswitch.org/>
>>             http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>>             http://www.cluecon.com <http://www.cluecon.com/>
>>
>>             Join Us At ClueCon - Aug 7-9, 2012
>>
>>             FreeSWITCH-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/>
>>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.com
>>
>>         Join Us At ClueCon - Aug 7-9, 2012
>>
>>         FreeSWITCH-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
>>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.com
>>
>>         Join Us At ClueCon - Aug 7-9, 2012
>>
>>         FreeSWITCH-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
>>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.com
>>
>>         Join Us At ClueCon - Aug 7-9, 2012
>>
>>         FreeSWITCH-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
>>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.com
>>
>>         Join Us At ClueCon - Aug 7-9, 2012
>>
>>         FreeSWITCH-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
>>
>>
>>
>>
>>         _________________________________________________________________________
>>
>>         Professional FreeSWITCH Consulting Services:
>>
>>         consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>
>>         http://www.freeswitchsolutions.com
>>
>>           
>>
>>         
>>
>>         
>>
>>           
>>
>>         Official FreeSWITCH Sites
>>
>>         http://www.freeswitch.org
>>
>>         http://wiki.freeswitch.org
>>
>>         http://www.cluecon.com
>>
>>           
>>
>>         Join Us At ClueCon - Aug 7-9, 2012
>>
>>           
>>
>>         FreeSWITCH-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:502354a432765721010175!
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.com
>>
>>     Join Us At ClueCon - Aug 7-9, 2012
>>
>>     FreeSWITCH-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
>     @Peter,
>
>     thanks for all your explanation.
>
>     Br,
>     Miha
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     FreeSWITCH-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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun Aug 12 03:46:15 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 11 Aug 2012 18:46:15 -0500
Subject: [Freeswitch-users] Is authorizing presence subscriptions
	possible?
In-Reply-To: <1FA7AE64-E5CB-48FA-AAD5-F5118BD97C53@freeswitch.org>
References: <5024F755.8080003@dgt.com.pl>
	<D869AABB-A75F-4198-8638-B7E06287C146@freeswitch.org>
	<52148.193.34.0.63.1344671661.squirrel@mail.dgtronik.com.pl>
	<1FA7AE64-E5CB-48FA-AAD5-F5118BD97C53@freeswitch.org>
Message-ID: <CAKbxfG_goYj5w2PhVvZ4Rzz8UCgrLZOV6kLjpx+tPTEg8N6BEA@mail.gmail.com>

We currently have not authorization layer on presence.

On Sat, Aug 11, 2012 at 10:23 AM, Ken Rice <krice at freeswitch.org> wrote:
> i dont think that is possible as a per user scenario, but it may be possible via a per sofile profile... outside of that i would look at sponsoring one of the devs to add such a feature
>
> Ken
> Sent from my iPad
>
> On Aug 11, 2012, at 2:54 AM, "Jakub Nadolski" <jakub.nadolski at dgt.com.pl> wrote:
>
>> Yes, this is what I would like to do.
>>
>> Kuba
>>
>>>
>>> as in on a per user basis? ie user x can subscribe but user y cannot?
>>> Ken
>>> Sent from my iPad
>>>
>>> On Aug 10, 2012, at 6:58 AM, Jakub Nadolski <jakub.nadolski at dgt.com.pl>
>>> wrote:
>>>
>>>> Hello everyone,
>>>>
>>>> I have a question about presence in mod sofia. I use Aastra phones and
>>>> BLF function works fine. But now I need to authorize subcriptions. So I
>>>> need configuration in which not all users can subscribe and receive
>>>> state of the extension in NOTIFY messages. Is that possible and if so
>>>> how can I do that?
>>>>
>>>> Thanks in advance,
>>>> Kuba
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 bdfoster at endigotech.com  Sun Aug 12 04:50:17 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 11 Aug 2012 20:50:17 -0400
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAKbxfG8Wavpt7COjPxm9BGA7LV-8_NZBS7koakMNTiTUBE-4=A@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F15141094@cantor.std.visionutv.se>
	<50236299.4070403@softnet.si>
	<CAKV-jFyJEGW72UwZ8AGkKPL9uvoab3fOUb7y5_wTx6_suPT0JQ@mail.gmail.com>
	<CAKbxfG8Wavpt7COjPxm9BGA7LV-8_NZBS7koakMNTiTUBE-4=A@mail.gmail.com>
Message-ID: <CAKV-jFzmZQpUw6wWbsaDe-3Z0Q0J_0r3+fQV7dCbf_tg7hM3aA@mail.gmail.com>

Packages are the best way, provided we can keep up and stay organized. Ken,
I have no problems helping with packages but I would have to do a ton of
research to get to a stage to be useful. I've been needing to learn how to
do debian packaging and this could be my excuse to finally learn.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 11, 2012 3:13 PM, "Anthony Minessale" <anthony.minessale at gmail.com>
wrote:

> Based on the inexplicable issues in cent6 working on debian packages might
> be very fruitful.  Talk to krice at freeswitch.org
> On Aug 10, 2012 8:32 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>
>> I need to update the wiki, but the install script for Debian is at
>> github.com/bdfoster/fs-debian-installer. It still gets the latest code
>> on master. If you guys want to give me some feedback I can incorporate the
>> proper changes. I stopped working on it because most are using centos and I
>> am not going to maintain something no one uses. I have the time, I have the
>> energy, but I need to know what people would like to see. Our internal
>> script is based on this one but I cannot release those changes to the world.
>>
>> Brian Foster
>> Endigo Computer LLC
>>
>> Sent from a mobile device.
>> On Aug 9, 2012 3:12 AM, "Miha" <miha at softnet.si> wrote:
>>
>>>  On 8/9/2012 8:36 AM, Peter Olsson wrote:
>>>
>>>  If you change to branch ?v1.2.stable?, the version will be 1.2.0.****
>>>
>>> ** **
>>>
>>> The core devs have not yet sent out any official instructions, but I
>>> guess that if you want to continue to follow the main development (with
>>> sometimes unstable code), just continue to do as you do today. If you want
>>> to stick to the 1.2 version, and only get bugfixes etc. for that version
>>> (but no new features), change to the branch v1.2.stable.****
>>>
>>> ** **
>>>
>>> I think there will be more official instructions soon, but this is that
>>> way I think it?s meant to work.****
>>>
>>> ** **
>>>
>>> /Peter****
>>>
>>> ** **
>>>
>>> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [
>>> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
>>> *F?r *Miha
>>> *Skickat:* den 9 augusti 2012 08:15
>>> *Till:* FreeSWITCH Users Help
>>> *?mne:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2****
>>>
>>> ** **
>>>
>>> Hi,
>>>
>>> so my current version is 1.2 rc2. After make current I will have 1.2? I
>>> try it on test server but still FreeSWITCH Version
>>> 1.2.0-rc2+git~20120809T022906Z~3389f32363 (1.2.0-rc2; git at commit
>>> 3389f32363 on Thu, 09 Aug 2012 02:29:06 Z)
>>>
>>>
>>> Thanks!
>>>
>>> Miha
>>>
>>> On 8/8/2012 5:02 PM, Ken Rice wrote:****
>>>
>>> the Primary repo is git.freeswitch.org/freeswitch.git this is the
>>> master repo, however, the github.com/freeswitch/freeswitch is an
>>> unofficial mirror...  updates to primary git repo should be pushed
>>> including tags and branches but we do not support those that wish to use
>>> other than official repos ****
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> K****
>>>
>>> ** **
>>>
>>> On Wed, Aug 8, 2012 at 8:39 AM, Brian Foster <bdfoster at endigotech.com>
>>> wrote:****
>>>
>>> Ah, looks like they publish to github as well. Afaik the primary repo is
>>> git.freeswitch.org, and that is where I get everything. ****
>>>
>>> Brian Foster
>>> Endigo Computer LLC****
>>>
>>> Sent from a mobile device.****
>>>
>>> On Aug 8, 2012 8:41 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:****
>>>
>>> What is this then? https://github.com/freeswitch/freeswitch ****
>>>
>>>
>>> Regards,
>>> Ben Langfeld
>>>
>>> ****
>>>
>>> On 8 August 2012 12:53, Brian Foster <bdfoster at endigotech.com> wrote:***
>>> *
>>>
>>> FreeSWITCH isn't published on GitHub.****
>>>
>>> Brian Foster
>>> Endigo Computer LLC****
>>>
>>> Sent from a mobile device.****
>>>
>>> On Aug 8, 2012 3:14 AM, "Stanislav Sinyagin" <ssinyagin at yahoo.com>
>>> wrote:****
>>>
>>> Ken,****
>>>
>>> care to push tags to Github as well?****
>>>
>>> ** **
>>>   ------------------------------
>>>
>>> *From:* Ken Rice <krice at freeswitch.org>
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Sent:* Tuesday, August 7, 2012 8:31 PM
>>> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2****
>>>
>>> ** **
>>>
>>> yes its there... git pull --tags****
>>>
>>> ** **
>>>
>>> On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net> wrote:**
>>> **
>>>
>>> I also checked tags, but didn't see it. Are you sure your branch is
>>> tracked upstream?****
>>>
>>>  ****
>>>
>>> --Dave****
>>>
>>> ** **
>>>  ------------------------------
>>>
>>> *From:* Ken Rice [mailto:krice at freeswitch.org]
>>> *To:* FreeSWITCH Users Help [mailto:
>>> freeswitch-users at lists.freeswitch.org]****
>>>
>>> *Sent:* Tue, 07 Aug 2012 11:15:50 -0700
>>> *Subject:* Re: [Freeswitch-users] Announcing FreeSWITCH 1.2 ****
>>>
>>>
>>>
>>> its Tagged, right now, we will be branching it shortly for tracking
>>> patches. Please note we had a couple of good users point out some smallish
>>> bugs so we're int he process of fixing those! so guys if you find anything
>>> wrong please open a jira, if you are at cluecon, find me and let me know
>>> the bug number ****
>>>
>>> ** **
>>>
>>> K****
>>>
>>> On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net> wrote:**
>>> **
>>>
>>> Where is the branch so I can checkout local?****
>>>
>>>  ****
>>>
>>> D:\fsnew>git branch -a
>>> * master
>>>   remotes/origin/FS-3432
>>>   remotes/origin/FS-4062
>>>   remotes/origin/HEAD -> origin/master
>>>   remotes/origin/dingaling_video
>>>   remotes/origin/master
>>>   remotes/origin/smgfs
>>>   remotes/origin/stable-test/freeswitch-1.2
>>>   remotes/origin/swk/fs_test
>>>   remotes/test/master****
>>>
>>> --Dave****
>>>
>>> ** **
>>>  ------------------------------
>>>
>>> *From:* Ken Rice [mailto:krice at freeswitch.org]
>>> *To:* FreeSWITCH Users Help [mailto:
>>> freeswitch-users at lists.freeswitch.org],
>>> freeswitch-dev at lists.freeswitch.org
>>> *Sent:* Tue, 07 Aug 2012 09:03:34 -0700 ****
>>>
>>>
>>> *Subject:* [Freeswitch-users] Announcing FreeSWITCH 1.2****
>>>
>>> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0! ** **
>>>
>>> ** **
>>>
>>> Get your copy today at
>>> http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 ! ****
>>>
>>> ** **
>>>
>>> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we
>>> will continue to fix bugs and security issues. giving you a stable platform
>>> for at least one year.****
>>>
>>> ** **
>>>
>>> Grab it today!****
>>>
>>> ** **
>>>
>>> The FreeSWITCH Team ****
>>>
>>>   ****
>>>
>>>  ****
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com/
>>>
>>> 
>>> /
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org/
>>> http://wiki.freeswitch.org/
>>> http://www.cluecon.com/
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org****
>>>
>>> ** **
>>>
>>>   ****
>>>
>>>  ****
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org****
>>>
>>> ** **
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> ****
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org****
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org****
>>>
>>> ** **
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org****
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org****
>>>
>>> ** **
>>>
>>>
>>>
>>>
>>> ****
>>>
>>> _________________________________________________________________________****
>>>
>>> Professional FreeSWITCH Consulting Services:****
>>>
>>> consulting at freeswitch.org****
>>>
>>> http://www.freeswitchsolutions.com****
>>>
>>> ** **
>>>
>>> ****
>>>
>>> ****
>>>
>>> ** **
>>>
>>> Official FreeSWITCH Sites****
>>>
>>> http://www.freeswitch.org****
>>>
>>> http://wiki.freeswitch.org****
>>>
>>> http://www.cluecon.com****
>>>
>>> ** **
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012****
>>>
>>> ** **
>>>
>>> FreeSWITCH-users mailing list****
>>>
>>> FreeSWITCH-users at 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:502354a432765721010175! ****
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> 
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> 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
>>>
>>>  @Peter,
>>>
>>> thanks for all your explanation.
>>>
>>> Br,
>>> Miha
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gerald.weber at besharp.at  Sun Aug 12 14:28:00 2012
From: gerald.weber at besharp.at (Gerald Weber)
Date: Sun, 12 Aug 2012 10:28:00 +0000
Subject: [Freeswitch-users] sofia_contact result
In-Reply-To: <CAKbxfG8u7C8S-yXTDxUf8a_gLNjh3biv9N2m5Zv=i3DiwvpnCQ@mail.gmail.com>
References: <D1925DECD316894DAADB84076C1B7D827E279E31@exchange.besharp.at>
	<CAKbxfG_bfFM9+drBr9oHQpXeFUzWfhCrEECkx0_Lt=K21VU9uQ@mail.gmail.com>
	<D1925DECD316894DAADB84076C1B7D827E279E5A@exchange.besharp.at>
	<CAKbxfG8u7C8S-yXTDxUf8a_gLNjh3biv9N2m5Zv=i3DiwvpnCQ@mail.gmail.com>
Message-ID: <D1925DECD316894DAADB84076C1B7D827E28440B@exchange.besharp.at>

I've build the keypress event into mod_snom, jira FS-4502.
Maybe someone find it useful :)

gw


-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Montag, 30. Juli 2012 17:26
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] sofia_contact result

its basically "sofia/" + <profile name> + "/" <contact provided from register>


On Mon, Jul 30, 2012 at 8:25 AM, Gerald Weber <gerald.weber at besharp.at> wrote:
> With rely i mean, ist he host/ip always between '@' and ':' or are there any other combinations ?
> sofia/<profile>/sip:<registered_user>@<ip>:<port>
>
> Background:
> I'm currenlty working on mod_snom to implement the KEYPRESS feature.
> To do so, i need the ip of the user, 2 choices came on my mind:
>
> 1. use sofia_contact as suggested by Brian Foster and parse the output.
> 2. do a select network_ip from registrations where reg_user = 'xxxx'
>
> sofia_contact looks better because i can pass the whole contact string from mod_callcenter using switch_api_execute without all the fiddling with domain, user, etc.
>
> Thats why i'm interested in the structure.
>
> ps: I created a feature request at snom to support notify talk last 
> november, but still no response :(
>
>
>
> -----Urspr?ngliche Nachricht-----
> Von: freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von 
> Anthony Minessale
> Gesendet: Montag, 30. Juli 2012 15:11
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] sofia_contact result
>
> Depends on what you mean by rely?
>
>
> On Mon, Jul 30, 2012 at 7:56 AM, Gerald Weber <gerald.weber at besharp.at> wrote:
>> Hi,
>>
>>
>>
>> can i rely on the structure of the result string i get back from e.g.:
>>
>>
>>
>> freeswitch at default> sofia_contact 2000 (or sofia_contact user/2000)
>>
>> sofia/internal/sip:2000 at 192.168.20.219:5060
>>
>>
>>
>> thanks
>>
>> gw
>>
>>
>> _____________________________________________________________________
>> _ ___ Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>  
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> e
>> rs
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org



--
Anthony Minessale II

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

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

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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


From wingcomm at hotmail.com  Sun Aug 12 12:44:03 2012
From: wingcomm at hotmail.com (R W)
Date: Sun, 12 Aug 2012 04:44:03 -0400
Subject: [Freeswitch-users] TLS on FreeSwitch not Working
In-Reply-To: <CALn233rwaSGbtbcOV8sJPRA561T9MoWooBRYk4E=RNuvq9UM+Q@mail.gmail.com>
References: <BAY156-W11F916CCDCFC6B2A6B3E2DD4CC0@phx.gbl>,
	<CALn233rwaSGbtbcOV8sJPRA561T9MoWooBRYk4E=RNuvq9UM+Q@mail.gmail.com>
Message-ID: <BAY156-W466967C2B4CE10A8B3AE2CD4B10@phx.gbl>


Mitch,
Thank you! I enabled logging and found my CA cert file misnamed. Re-naming it fixed the problem!
-Rob

> From: mitch.capper at gmail.com
> Date: Thu, 9 Aug 2012 09:33:28 -0700
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] TLS on FreeSwitch not Working
> 
> Turn on sofia tport logging it will tell you what its unable to setup
> the TLS connection.
> 
> ~Mitch
> 
> On Wed, Aug 8, 2012 at 10:09 PM, R W <wingcomm at hotmail.com> wrote:
> > Hi All,
> >
> > I cannot seem to get TLS running on the sofia "internal" profile. Any
> > assistance would be appreciated.
> >
> > I'm running FreeSWITCH Version 1.2.0-rc2+git~20120808T025758Z~9ac586adc8
> > (1.2.0-rc2; git at commit 9ac586adc8 on Wed, 08 Aug 2012 02:57:58 Z) on
> > Ubuntu 12.04 LTS.
> >
> > When I set internal_ssl_enable=true, and reload the sofia internal profile,
> > I get the "usual" error:
> >
> > 2012-08-09 00:34:14.174431 [ERR] sofia.c:2289 Error Creating SIP UA for
> > profile: internal
> >
> > <!-- SIP Profile: Internal -->
> > <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=true"/>
> > <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>
> >
> > I verified that the OpenSSL development libraries were installed (Ubuntu
> > package libssl-dev) and looked for references to ssl in the output from the
> > compilation process and saw this:
> >
> > "checking for openssl... yes
> >
> > checking openssl_CFLAGS...
> >
> > checking openssl_LIBS... -lssl -lcrypto
> >
> >   adding "-DHAVE_OPENSSL" to SWITCH_AM_CFLAGS"
> >
> >
> > ...
> >
> >
> > "checking OpenSSL options with pkg-config... found
> >
> > checking for gdi32... no
> >
> > checking for CRYPTO_lock in -lcrypto... yes
> >
> > checking for SSL_connect in -lssl... yes
> >
> > checking openssl/x509.h usability... yes
> >
> > checking openssl/x509.h presence... yes
> >
> > checking for openssl/x509.h... yes
> >
> > checking openssl/rsa.h usability... yes
> >
> > checking openssl/rsa.h presence... yes
> >
> > checking for openssl/rsa.h... yes
> >
> > checking openssl/crypto.h usability... yes
> >
> > checking openssl/crypto.h presence... yes
> >
> > checking for openssl/crypto.h... yes
> >
> > checking openssl/pem.h usability... yes
> >
> > checking openssl/pem.h presence... yes
> >
> > checking for openssl/pem.h... yes
> >
> > checking openssl/ssl.h usability... yes
> >
> > checking openssl/ssl.h presence... yes
> >
> > checking for openssl/ssl.h... yes
> >
> > checking openssl/err.h usability... yes
> >
> > checking openssl/err.h presence... yes
> >
> > checking for openssl/err.h... yes
> >
> > checking openssl/pkcs12.h usability... yes
> >
> > checking openssl/pkcs12.h presence... yes
> >
> > checking for openssl/pkcs12.h... yes
> >
> > checking for ENGINE_init... yes
> >
> > checking openssl/engine.h usability... yes
> >
> > checking openssl/engine.h presence... yes
> >
> > checking for openssl/engine.h... yes
> >
> > checking for ENGINE_load_builtin_engines... yes
> >
> > checking for RAND_status... yes
> >
> > checking for RAND_screen... no
> >
> > checking for RAND_egd... yes
> >
> > checking for CRYPTO_cleanup_all_ex_data... yes
> >
> > checking for "/dev/urandom"... yes
> >
> > checking CA cert bundle install path...
> > ${prefix}/share/curl/curl-ca-bundle.crt
> >
> > checking for inflateEnd in -lz... yes"
> >
> >
> > Is there anything else I should be checking. Does freeswitch send logs
> > anywhere other than ../freeswitch/log/ ?
> >
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
-------------- next part --------------
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From admin at blindi.net  Mon Aug 13 01:44:33 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Sun, 12 Aug 2012 23:44:33 +0200 (CEST)
Subject: [Freeswitch-users] lua problem: functions in streamfile pause,
 seek not working on dtmf digit
In-Reply-To: <1FA7AE64-E5CB-48FA-AAD5-F5118BD97C53@freeswitch.org>
References: <5024F755.8080003@dgt.com.pl>
	<D869AABB-A75F-4198-8638-B7E06287C146@freeswitch.org>
	<52148.193.34.0.63.1344671661.squirrel@mail.dgtronik.com.pl>
	<1FA7AE64-E5CB-48FA-AAD5-F5118BD97C53@freeswitch.org>
Message-ID: <Pine.LNX.4.64.1208122330340.14631@1a-8140.antagus.de>


Hi all, i have 2 solutions:
i like to 1000 ms backword in a soundfile, after pressing the dtmf digit 
1.
my 1. script:

function read_msg_menu( )
  session:setInputCallback("onInput");
session:streamFile("/home/sounds/podcast.wav")
end

function onInput(s, type, obj)
  if (type == "dtmf") then
elseif (obj.digit   == "1")  then
return("seek:+1000");

elseif (obj.digit   == "3")  then
return("seek:-1000");

end

not working. streamfile ignores all dtmf inputs.

solution 2:
  session:streamFile("/home/sounds/podcast1.wav", onInput)

not working. Streamfile breaks.

Can you help plese?
thanks


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From th982a at googlemail.com  Mon Aug 13 02:04:18 2012
From: th982a at googlemail.com (Tamer Higazi)
Date: Mon, 13 Aug 2012 00:04:18 +0200
Subject: [Freeswitch-users] make fs listen to internal ip-adress
Message-ID: <50282862.2090200@googlemail.com>

Hi people!
I have the problem making freeswitch listen to an internal IP-Adress,
that trunks in the internal network, connected at a specific NIC board,
are able to register.

eth0 is connected to a DSL modem, eth3 my ATA Adapter, with 2 Trunks
that should register / login at freeswitch.


in the CLI, "sofia status" tells me, that "internal", is the WAN-IPadress:

internal    profile             sip:mod_sofia at 212.255.33.187:5060   
RUNNING (0)


what I of course don't like.

if I set in the internal.xml,

<param name="rtp-ip" value="192.168.2.10"/>
<param name="sip-ip" value="192.168.2.10"/>

I get the error on the console:

here is the error: 2012-08-12 23:42:12.107512 [ERR] sofia.c:2289 Error
Creating SIP UA for profile: internal

How do I change the setting in "internal.xml" proparly, so that FS
listens to everything in 192.168.2.10....



I would kindly thank you



Tamer


From curriegrad2004 at gmail.com  Mon Aug 13 04:37:29 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 12 Aug 2012 17:37:29 -0700
Subject: [Freeswitch-users] make fs listen to internal ip-adress
In-Reply-To: <50282862.2090200@googlemail.com>
References: <50282862.2090200@googlemail.com>
Message-ID: <CAMWiMy_bEnMJ1f=XFiZ=OPKm21tCaKBGnFRfT2oxrLvDB4k1HQ@mail.gmail.com>

Are you sure there's another profile that isn't listening on to the
same ports where your "internal" profile is trying to bind to?

On Sun, Aug 12, 2012 at 3:04 PM, Tamer Higazi <th982a at googlemail.com> wrote:
> Hi people!
> I have the problem making freeswitch listen to an internal IP-Adress,
> that trunks in the internal network, connected at a specific NIC board,
> are able to register.
>
> eth0 is connected to a DSL modem, eth3 my ATA Adapter, with 2 Trunks
> that should register / login at freeswitch.
>
>
> in the CLI, "sofia status" tells me, that "internal", is the WAN-IPadress:
>
> internal    profile             sip:mod_sofia at 212.255.33.187:5060
> RUNNING (0)
>
>
> what I of course don't like.
>
> if I set in the internal.xml,
>
> <param name="rtp-ip" value="192.168.2.10"/>
> <param name="sip-ip" value="192.168.2.10"/>
>
> I get the error on the console:
>
> here is the error: 2012-08-12 23:42:12.107512 [ERR] sofia.c:2289 Error
> Creating SIP UA for profile: internal
>
> How do I change the setting in "internal.xml" proparly, so that FS
> listens to everything in 192.168.2.10....
>
>
>
> I would kindly thank you
>
>
>
> Tamer
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From admin at blindi.net  Mon Aug 13 05:37:23 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Mon, 13 Aug 2012 03:37:23 +0200 (CEST)
Subject: [Freeswitch-users] fs crashs after more then 100 dtmf events on a
	call
In-Reply-To: <5026C7D8.80008@quentustech.com>
References: <1FFF97C269757C458224B7C895F35F15141094@cantor.std.visionutv.se>
	<50236299.4070403@softnet.si>
	<CAKV-jFyJEGW72UwZ8AGkKPL9uvoab3fOUb7y5_wTx6_suPT0JQ@mail.gmail.com>
	<5026C7D8.80008@quentustech.com>
Message-ID: <Pine.LNX.4.64.1208130308060.16739@1a-8140.antagus.de>

Hi all, i have writen a podcast system in lua.
After more then 100 dtmf input callback events fs crashs.
my luacode:
-- read potcast menu
function read_msg_menu( )
  session:setInputCallback("onInput");
digits =
session:playAndGetDigits(1, 1, 3, 5000, "#",
"/usr/local/freeswitch/sounds/dorf/msgs_menu.alaw",
"/usr/local/freeswitch/sounds/ivr/falsche-eingabe.wav",  "[*01-9]")
  if (digits  == "1")  then
-- set the pointe to - 1 by pressing 1
msg_pointer_minus(1)
  msg_file_play =
  api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
  session:streamFile(msg_file_play)
read_msg_menu()
  end

if  (digits  == "3")  then
-- forward + 1 
msg_pointer(1)

msg_file_play =
api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
  session:streamFile(msg_file_play)
read_msg_menu()
  end
end
-- input callback
function onInput(s, type, obj)
if (obj.digit   == "1")  then
  msg_pointer_minus(1)
  msg_file_play =
  api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
   session:streamFile(msg_file_play)
  return("false");
end

if (obj.digit   == "3")  then
  msg_pointer(1)
  msg_file_play =
  api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
   session:streamFile(msg_file_play)
  return("false");
end
end

I play from next to next podcast
the inputcallback catch all dtmf tones correctly.
I terminate the playback before goto the next entry.

fs crashs now:
for example:
I have 200 podcastfiles, and i pressing 3 3 3 3 and so on, to skip the 
long time of a playback.
Abbout 100 navigation entrys, fs crash.
fs_cli is dead.

what is the problem please?
can your help me please to creating a jira?

thanks



---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From covici at ccs.covici.com  Mon Aug 13 07:34:31 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 12 Aug 2012 23:34:31 -0400
Subject: [Freeswitch-users] fs crashs after more then 100 dtmf events on
	a call
In-Reply-To: <Pine.LNX.4.64.1208130308060.16739@1a-8140.antagus.de>
References: <1FFF97C269757C458224B7C895F35F15141094@cantor.std.visionutv.se>
	<50236299.4070403@softnet.si>
	<CAKV-jFyJEGW72UwZ8AGkKPL9uvoab3fOUb7y5_wTx6_suPT0JQ@mail.gmail.com>
	<5026C7D8.80008@quentustech.com>
	<Pine.LNX.4.64.1208130308060.16739@1a-8140.antagus.de>
Message-ID: <1676.1344828871@ccs.covici.com>

Not sure that is a good design, but I think you are running in to the
maximum number of forwards.  There is a variable for that, check the
wiki.

Thomas Hoellriegel <admin at blindi.net> wrote:

> Hi all, i have writen a podcast system in lua.
> After more then 100 dtmf input callback events fs crashs.
> my luacode:
> -- read potcast menu
> function read_msg_menu( )
>  session:setInputCallback("onInput");
> digits =
> session:playAndGetDigits(1, 1, 3, 5000, "#",
> "/usr/local/freeswitch/sounds/dorf/msgs_menu.alaw",
> "/usr/local/freeswitch/sounds/ivr/falsche-eingabe.wav",  "[*01-9]")
>  if (digits  == "1")  then
> -- set the pointe to - 1 by pressing 1
> msg_pointer_minus(1)
>  msg_file_play =
>  api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
>  session:streamFile(msg_file_play)
> read_msg_menu()
>  end
> 
> if  (digits  == "3")  then
> -- forward + 1 msg_pointer(1)
> 
> msg_file_play =
> api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
>  session:streamFile(msg_file_play)
> read_msg_menu()
>  end
> end
> -- input callback
> function onInput(s, type, obj)
> if (obj.digit   == "1")  then
>  msg_pointer_minus(1)
>  msg_file_play =
>  api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
>   session:streamFile(msg_file_play)
>  return("false");
> end
> 
> if (obj.digit   == "3")  then
>  msg_pointer(1)
>  msg_file_play =
>  api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
>   session:streamFile(msg_file_play)
>  return("false");
> end
> end
> 
> I play from next to next podcast
> the inputcallback catch all dtmf tones correctly.
> I terminate the playback before goto the next entry.
> 
> fs crashs now:
> for example:
> I have 200 podcastfiles, and i pressing 3 3 3 3 and so on, to skip the
> long time of a playback.
> Abbout 100 navigation entrys, fs crash.
> fs_cli is dead.
> 
> what is the problem please?
> can your help me please to creating a jira?
> 
> thanks
> 
> 
> 
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 krice at freeswitch.org  Mon Aug 13 08:53:16 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 12 Aug 2012 23:53:16 -0500
Subject: [Freeswitch-users] fs crashs after more then 100 dtmf events on
	a call
In-Reply-To: <1676.1344828871@ccs.covici.com>
References: <1FFF97C269757C458224B7C895F35F15141094@cantor.std.visionutv.se>
	<50236299.4070403@softnet.si>
	<CAKV-jFyJEGW72UwZ8AGkKPL9uvoab3fOUb7y5_wTx6_suPT0JQ@mail.gmail.com>
	<5026C7D8.80008@quentustech.com>
	<Pine.LNX.4.64.1208130308060.16739@1a-8140.antagus.de>
	<1676.1344828871@ccs.covici.com>
Message-ID: <2C3B8DC9-5E1F-4497-947F-B653A47E60A8@freeswitch.org>

get a back trace and open a jira.. also if you can atach the script you are using to the jira... if its just hanging use gcore to get the coredump then follow the rest of the back trace guidelines on the wiki

Ken
Sent from my iPad

On Aug 12, 2012, at 10:34 PM, covici at ccs.covici.com wrote:

> Not sure that is a good design, but I think you are running in to the
> maximum number of forwards.  There is a variable for that, check the
> wiki.
> 
> Thomas Hoellriegel <admin at blindi.net> wrote:
> 
>> Hi all, i have writen a podcast system in lua.
>> After more then 100 dtmf input callback events fs crashs.
>> my luacode:
>> -- read potcast menu
>> function read_msg_menu( )
>> session:setInputCallback("onInput");
>> digits =
>> session:playAndGetDigits(1, 1, 3, 5000, "#",
>> "/usr/local/freeswitch/sounds/dorf/msgs_menu.alaw",
>> "/usr/local/freeswitch/sounds/ivr/falsche-eingabe.wav",  "[*01-9]")
>> if (digits  == "1")  then
>> -- set the pointe to - 1 by pressing 1
>> msg_pointer_minus(1)
>> msg_file_play =
>> api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
>> session:streamFile(msg_file_play)
>> read_msg_menu()
>> end
>> 
>> if  (digits  == "3")  then
>> -- forward + 1 msg_pointer(1)
>> 
>> msg_file_play =
>> api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
>> session:streamFile(msg_file_play)
>> read_msg_menu()
>> end
>> end
>> -- input callback
>> function onInput(s, type, obj)
>> if (obj.digit   == "1")  then
>> msg_pointer_minus(1)
>> msg_file_play =
>> api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
>>  session:streamFile(msg_file_play)
>> return("false");
>> end
>> 
>> if (obj.digit   == "3")  then
>> msg_pointer(1)
>> msg_file_play =
>> api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
>>  session:streamFile(msg_file_play)
>> return("false");
>> end
>> end
>> 
>> I play from next to next podcast
>> the inputcallback catch all dtmf tones correctly.
>> I terminate the playback before goto the next entry.
>> 
>> fs crashs now:
>> for example:
>> I have 200 podcastfiles, and i pressing 3 3 3 3 and so on, to skip the
>> long time of a playback.
>> Abbout 100 navigation entrys, fs crash.
>> fs_cli is dead.
>> 
>> what is the problem please?
>> can your help me please to creating a jira?
>> 
>> thanks
>> 
>> 
>> 
>> ---------------
>> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
>> http://www.blindi.net/callback
>> homepage: http://www.blindi.net
>> blinde-misc mailingliste f?r blinde. anmeldung unter:
>> http://www.blindi.net/mailman/listinfo/blinde-misc
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From covici at ccs.covici.com  Mon Aug 13 11:38:21 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 13 Aug 2012 03:38:21 -0400
Subject: [Freeswitch-users] fs crashs after more then 100 dtmf events on
	a call
In-Reply-To: <2C3B8DC9-5E1F-4497-947F-B653A47E60A8@freeswitch.org>
References: <1FFF97C269757C458224B7C895F35F15141094@cantor.std.visionutv.se>
	<50236299.4070403@softnet.si>
	<CAKV-jFyJEGW72UwZ8AGkKPL9uvoab3fOUb7y5_wTx6_suPT0JQ@mail.gmail.com>
	<5026C7D8.80008@quentustech.com>
	<Pine.LNX.4.64.1208130308060.16739@1a-8140.antagus.de>
	<1676.1344828871@ccs.covici.com>
	<2C3B8DC9-5E1F-4497-947F-B653A47E60A8@freeswitch.org>
Message-ID: <2897.1344843501@ccs.covici.com>

OK, here I go showing my ignorance -- what is gcore --sounds like it
could be quite useful, but I don't seem to have it in the repository,
except as a package manager!

Ken Rice <krice at freeswitch.org> wrote:

> get a back trace and open a jira.. also if you can atach the script you are using to the jira... if its just hanging use gcore to get the coredump then follow the rest of the back trace guidelines on the wiki
> 
> Ken
> Sent from my iPad
> 
> On Aug 12, 2012, at 10:34 PM, covici at ccs.covici.com wrote:
> 
> > Not sure that is a good design, but I think you are running in to the
> > maximum number of forwards.  There is a variable for that, check the
> > wiki.
> > 
> > Thomas Hoellriegel <admin at blindi.net> wrote:
> > 
> >> Hi all, i have writen a podcast system in lua.
> >> After more then 100 dtmf input callback events fs crashs.
> >> my luacode:
> >> -- read potcast menu
> >> function read_msg_menu( )
> >> session:setInputCallback("onInput");
> >> digits =
> >> session:playAndGetDigits(1, 1, 3, 5000, "#",
> >> "/usr/local/freeswitch/sounds/dorf/msgs_menu.alaw",
> >> "/usr/local/freeswitch/sounds/ivr/falsche-eingabe.wav",  "[*01-9]")
> >> if (digits  == "1")  then
> >> -- set the pointe to - 1 by pressing 1
> >> msg_pointer_minus(1)
> >> msg_file_play =
> >> api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
> >> session:streamFile(msg_file_play)
> >> read_msg_menu()
> >> end
> >> 
> >> if  (digits  == "3")  then
> >> -- forward + 1 msg_pointer(1)
> >> 
> >> msg_file_play =
> >> api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
> >> session:streamFile(msg_file_play)
> >> read_msg_menu()
> >> end
> >> end
> >> -- input callback
> >> function onInput(s, type, obj)
> >> if (obj.digit   == "1")  then
> >> msg_pointer_minus(1)
> >> msg_file_play =
> >> api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
> >>  session:streamFile(msg_file_play)
> >> return("false");
> >> end
> >> 
> >> if (obj.digit   == "3")  then
> >> msg_pointer(1)
> >> msg_file_play =
> >> api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
> >>  session:streamFile(msg_file_play)
> >> return("false");
> >> end
> >> end
> >> 
> >> I play from next to next podcast
> >> the inputcallback catch all dtmf tones correctly.
> >> I terminate the playback before goto the next entry.
> >> 
> >> fs crashs now:
> >> for example:
> >> I have 200 podcastfiles, and i pressing 3 3 3 3 and so on, to skip the
> >> long time of a playback.
> >> Abbout 100 navigation entrys, fs crash.
> >> fs_cli is dead.
> >> 
> >> what is the problem please?
> >> can your help me please to creating a jira?
> >> 
> >> thanks
> >> 
> >> 
> >> 
> >> ---------------
> >> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> >> http://www.blindi.net/callback
> >> homepage: http://www.blindi.net
> >> blinde-misc mailingliste f?r blinde. anmeldung unter:
> >> http://www.blindi.net/mailman/listinfo/blinde-misc
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >> 
> >> 
> >> 
> >> 
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >> 
> >> Join Us At ClueCon - Aug 7-9, 2012
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > Join Us At ClueCon - Aug 7-9, 2012
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 peter.olsson at visionutveckling.se  Mon Aug 13 11:50:17 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 13 Aug 2012 07:50:17 +0000
Subject: [Freeswitch-users] fs crashs after more then 100 dtmf events
	on	a call
Message-ID: <1FFF97C269757C458224B7C895F35F1514327E@cantor.std.visionutv.se>

Please read this, it will describe most of it: http://wiki.freeswitch.org/wiki/Reporting_Bugs

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r covici at ccs.covici.com
Skickat: den 13 augusti 2012 09:38
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] fs crashs after more then 100 dtmf events on a call

OK, here I go showing my ignorance -- what is gcore --sounds like it could be quite useful, but I don't seem to have it in the repository, except as a package manager!

Ken Rice <krice at freeswitch.org> wrote:

> get a back trace and open a jira.. also if you can atach the script 
> you are using to the jira... if its just hanging use gcore to get the 
> coredump then follow the rest of the back trace guidelines on the wiki
> 
> Ken
> Sent from my iPad
> 
> On Aug 12, 2012, at 10:34 PM, covici at ccs.covici.com wrote:
> 
> > Not sure that is a good design, but I think you are running in to 
> > the maximum number of forwards.  There is a variable for that, check 
> > the wiki.
> > 
> > Thomas Hoellriegel <admin at blindi.net> wrote:
> > 
> >> Hi all, i have writen a podcast system in lua.
> >> After more then 100 dtmf input callback events fs crashs.
> >> my luacode:
> >> -- read potcast menu
> >> function read_msg_menu( )
> >> session:setInputCallback("onInput");
> >> digits =
> >> session:playAndGetDigits(1, 1, 3, 5000, "#", 
> >> "/usr/local/freeswitch/sounds/dorf/msgs_menu.alaw",
> >> "/usr/local/freeswitch/sounds/ivr/falsche-eingabe.wav",  "[*01-9]") 
> >> if (digits  == "1")  then
> >> -- set the pointe to - 1 by pressing 1
> >> msg_pointer_minus(1)
> >> msg_file_play =
> >> api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
> >> session:streamFile(msg_file_play)
> >> read_msg_menu()
> >> end
> >> 
> >> if  (digits  == "3")  then
> >> -- forward + 1 msg_pointer(1)
> >> 
> >> msg_file_play =
> >> api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
> >> session:streamFile(msg_file_play)
> >> read_msg_menu()
> >> end
> >> end
> >> -- input callback
> >> function onInput(s, type, obj)
> >> if (obj.digit   == "1")  then
> >> msg_pointer_minus(1)
> >> msg_file_play =
> >> api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
> >>  session:streamFile(msg_file_play)
> >> return("false");
> >> end
> >> 
> >> if (obj.digit   == "3")  then
> >> msg_pointer(1)
> >> msg_file_play =
> >> api:execute("hash", "select/mailbox_pointer_msg_file/" ..mailbox);
> >>  session:streamFile(msg_file_play)
> >> return("false");
> >> end
> >> end
> >> 
> >> I play from next to next podcast
> >> the inputcallback catch all dtmf tones correctly.
> >> I terminate the playback before goto the next entry.
> >> 
> >> fs crashs now:
> >> for example:
> >> I have 200 podcastfiles, and i pressing 3 3 3 3 and so on, to skip 
> >> the long time of a playback.
> >> Abbout 100 navigation entrys, fs crash.
> >> fs_cli is dead.
> >> 
> >> what is the problem please?
> >> can your help me please to creating a jira?
> >> 
> >> thanks
> >> 
> >> 
> >> 
> >> ---------------
> >> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> >> http://www.blindi.net/callback
> >> homepage: http://www.blindi.net
> >> blinde-misc mailingliste f?r blinde. anmeldung unter:
> >> http://www.blindi.net/mailman/listinfo/blinde-misc
> >> ___________________________________________________________________
> >> ______ Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >> 
> >>  
> >> 
> >> 
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >> 
> >> Join Us At ClueCon - Aug 7-9, 2012
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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
> > 
> > ____________________________________________________________________
> > _____ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> >  
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > Join Us At ClueCon - Aug 7-9, 2012
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u
> > sers
> > http://www.freeswitch.org
> 
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
>  
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org

--
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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at 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:5028add332761199412569!


From cristian.re.work at gmail.com  Mon Aug 13 13:05:53 2012
From: cristian.re.work at gmail.com (cristian re)
Date: Mon, 13 Aug 2012 11:05:53 +0200
Subject: [Freeswitch-users] originate in api_hangup_hook
Message-ID: <CABx_eV35LXsNwTvNpoWmaLTzuUqoQaPw2DiURH6OccTApZ2+6A@mail.gmail.com>

Hi,
I have a question about originate in api_hangup_hook.
I'm looking for call back the calling number after hang up, I made a test
in dialplan but if I make a call with a mobile phone the originate is so
fast that often goes to voicemail.
Is there a way to call back after some seconds?

My dialplan for this test is:

<context name="cb_test">
    <extension name="cb_test">
      <condition>
        <action application="pre_answer"/>
        <action application="set" data="api_hangup_hook=originate
sofia/external/${caller_id_number}@x.x.x.x CB000000"/>
        <action application="sleep" data="5000"/>
        <action application="hangup"/>
      </condition>
    </extension>
</context>
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From godson.g at gmail.com  Mon Aug 13 14:49:17 2012
From: godson.g at gmail.com (Godson Gera)
Date: Mon, 13 Aug 2012 16:19:17 +0530
Subject: [Freeswitch-users] originate in api_hangup_hook
In-Reply-To: <CABx_eV35LXsNwTvNpoWmaLTzuUqoQaPw2DiURH6OccTApZ2+6A@mail.gmail.com>
References: <CABx_eV35LXsNwTvNpoWmaLTzuUqoQaPw2DiURH6OccTApZ2+6A@mail.gmail.com>
Message-ID: <CAPTXd0rfBzB1-6vgE69UuVpNmsUqOj+=VWUsR6FD3kc-bPepsg@mail.gmail.com>

You can use sched_api to do that. There is an example here.

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

On Mon, Aug 13, 2012 at 2:35 PM, cristian re <cristian.re.work at gmail.com>wrote:

> Hi,
> I have a question about originate in api_hangup_hook.
> I'm looking for call back the calling number after hang up, I made a test
> in dialplan but if I make a call with a mobile phone the originate is so
> fast that often goes to voicemail.
> Is there a way to call back after some seconds?
>
> My dialplan for this test is:
>
> <context name="cb_test">
>     <extension name="cb_test">
>       <condition>
>         <action application="pre_answer"/>
>         <action application="set" data="api_hangup_hook=originate
> sofia/external/${caller_id_number}@x.x.x.x CB000000"/>
>         <action application="sleep" data="5000"/>
>         <action application="hangup"/>
>       </condition>
>     </extension>
> </context>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Thanks & Regards,
Godson Gera
SIP Consultant <http://godson.in>
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From cristian.re.work at gmail.com  Mon Aug 13 15:11:59 2012
From: cristian.re.work at gmail.com (cristian re)
Date: Mon, 13 Aug 2012 13:11:59 +0200
Subject: [Freeswitch-users] originate in api_hangup_hook
In-Reply-To: <CAPTXd0rfBzB1-6vgE69UuVpNmsUqOj+=VWUsR6FD3kc-bPepsg@mail.gmail.com>
References: <CABx_eV35LXsNwTvNpoWmaLTzuUqoQaPw2DiURH6OccTApZ2+6A@mail.gmail.com>
	<CAPTXd0rfBzB1-6vgE69UuVpNmsUqOj+=VWUsR6FD3kc-bPepsg@mail.gmail.com>
Message-ID: <CABx_eV27yqUNfvn_vj6wM93vsKN1BSdHWfuYWaXwH9jB32G_5w@mail.gmail.com>

Yes, it works.
Thanks



2012/8/13 Godson Gera <godson.g at gmail.com>

> You can use sched_api to do that. There is an example here.
>
> http://wiki.freeswitch.org/wiki/Mod_commands#sched_api
>
> On Mon, Aug 13, 2012 at 2:35 PM, cristian re <cristian.re.work at gmail.com>wrote:
>
>> Hi,
>> I have a question about originate in api_hangup_hook.
>> I'm looking for call back the calling number after hang up, I made a test
>> in dialplan but if I make a call with a mobile phone the originate is so
>> fast that often goes to voicemail.
>> Is there a way to call back after some seconds?
>>
>> My dialplan for this test is:
>>
>> <context name="cb_test">
>>     <extension name="cb_test">
>>       <condition>
>>         <action application="pre_answer"/>
>>         <action application="set" data="api_hangup_hook=originate
>> sofia/external/${caller_id_number}@x.x.x.x CB000000"/>
>>         <action application="sleep" data="5000"/>
>>         <action application="hangup"/>
>>       </condition>
>>     </extension>
>> </context>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Thanks & Regards,
> Godson Gera
> SIP Consultant <http://godson.in>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From eka at ekanet.net  Mon Aug 13 13:07:51 2012
From: eka at ekanet.net (Emrah)
Date: Mon, 13 Aug 2012 11:07:51 +0200
Subject: [Freeswitch-users] The domain_name variable and domains in
	FreeSWITCH
Message-ID: <20120813090751.GA15900@willow.kavun.ch>

Hello all,

I have been an Asterisk user for almost 10 years... God I'm glad I took the leap and tried FS!
Congratulations to all of you for the amazing dev you've put together.

I have been trying to understand in depth how the domain_name variable is used throughout FS. Or more in general, what defines a domain name and how is it used?

It doesn't seem that FS is domain sensitive for registrations / authenticated calls. If both sip.example.com and fs.example.com are IN A 1.2.3.4, I can register on both domains with no 
distinction whatsoever.

How does that work?

Best to you and thanks for your precious answers.
-- 
Emrah 
<eka at ekanet.net>


From rajkumar.kanniappan at sasken.com  Mon Aug 13 13:24:02 2012
From: rajkumar.kanniappan at sasken.com (Rajkumar Kanniappan)
Date: Mon, 13 Aug 2012 14:54:02 +0530
Subject: [Freeswitch-users] NORMAL CIRCUIT CONGESTION
Message-ID: <6F91E0FFDA542149961F7BDED2D2B94B568E5514D8@EXGMBX01.sasken.com>

Hi,

I had connected my FreeSWITCH server to ePBX using sanoma A102DE, and installed wanpipe drivers and loaded freetdm module with the below configurations.

When I tried to make an outgoing call to a PBX extension(PSTN)

I got the following error.
<application action="bridge" data="freetdm/wp1/16/1000"/>

2012-08-13 14:47:54.172568 [ERR] ftdm_io.c:1921 Failed to open channel 1:16
2012-08-13 14:47:54.172568 [NOTICE] mod_freetdm.c:1622 Close Channel N/A [CS_NEW]
2012-08-13 14:47:54.172568 [NOTICE] switch_ivr_originate.c:2544 Cannot create outgoing channel of type [freetdm] cause: [NORMAL_CIRCUIT_CONGESTION]

Please help me out in solving this problem.

My configuration files are,
[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE_API, Comment

[wanpipe1]
CARD_TYPE       = AFT
S514CPU         = A
CommPort        = PRI
AUTO_PCISLOT    = YES
PCISLOT         = 4
PCIBUS          = 2
FE_MEDIA        = E1
FE_LCODE        = HDB3
FE_FRAME        = CRC4
FE_LINE         = 1
TE_CLOCK        = NORMAL
TE_REF_CLOCK    = 0
TE_HIGHIMPEDANCE        = NO
TE_RX_SLEVEL    = 430
LBO             = 120OH
TE_SIG_MODE     = CCS
FE_TXTRISTATE   = NO
MTU             = 1500
UDPPORT         = 9000
TTL             = 255
IGNORE_FRONT_END = NO
TDMV_SPAN       = 1
TDMV_DCHAN      = 16
TDMV_HW_DTMF    = YES

[w1g1]
ACTIVE_CH       = ALL
TDMV_ECHO_OFF   = NO
TDMV_HWEC       = YES



freeswitch.conf.xml
<configuration name="freetdm.conf" description="Freetdm Configuration">
  <settings>
    <param name="debug" value="0"/>
  </settings>
                <config_profiles>
                <profile name=" cfg_prof">
                        <param name="switchtype" value="qsig" />
                        <param name="interface" value="pri_cpe"/>
                </profile>
                </config_profiles>
        <sangoma_pri_spans>
                <span name="wp1" cfgprofile="cfg_prof">
                        <param name="outbound-bc-user-layer1" value="ulaw"/>
                        <param name="send-cid-name" value="yes"/>
                        <param name="dialplan" value="XML"/>
                        <param name="context" value="PSTN"/>
                </span>
        </sangoma_pri_spans>
</configuration>

freetdm.conf
[span wanpipe wp1]
trunk_type => e1
group=grp1
b-channel => 1:1-15
b-channel => 1:17-31
d-channel => 1:16

Thanks
Rajkumar


________________________________
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From jakub.nadolski at dgt.com.pl  Mon Aug 13 13:25:36 2012
From: jakub.nadolski at dgt.com.pl (Jakub Nadolski)
Date: Mon, 13 Aug 2012 11:25:36 +0200
Subject: [Freeswitch-users] Is authorizing presence subscriptions
	possible?
In-Reply-To: <1FA7AE64-E5CB-48FA-AAD5-F5118BD97C53@freeswitch.org>
References: <5024F755.8080003@dgt.com.pl>
	<D869AABB-A75F-4198-8638-B7E06287C146@freeswitch.org>
	<52148.193.34.0.63.1344671661.squirrel@mail.dgtronik.com.pl>
	<1FA7AE64-E5CB-48FA-AAD5-F5118BD97C53@freeswitch.org>
Message-ID: <5028C810.9010103@dgt.com.pl>

W dniu 11.08.2012 17:23, Ken Rice pisze:
> i dont think that is possible as a per user scenario, but it may be possible via a per sofile profile... outside of that i would look at sponsoring one of the devs to add such a feature
>
> Ken
> Sent from my iPad
>
> On Aug 11, 2012, at 2:54 AM, "Jakub Nadolski" <jakub.nadolski at dgt.com.pl> wrote:
>
>> Yes, this is what I would like to do.
>>
>> Kuba
>>
>>> as in on a per user basis? ie user x can subscribe but user y cannot?
>>> Ken
>>> Sent from my iPad
>>>
>>> On Aug 10, 2012, at 6:58 AM, Jakub Nadolski <jakub.nadolski at dgt.com.pl>
>>> wrote:
>>>
>>>> Hello everyone,
>>>>
>>>> I have a question about presence in mod sofia. I use Aastra phones and
>>>> BLF function works fine. But now I need to authorize subcriptions. So I
>>>> need configuration in which not all users can subscribe and receive
>>>> state of the extension in NOTIFY messages. Is that possible and if so
>>>> how can I do that?
>>>>
>>>> Thanks in advance,
>>>> Kuba
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
Thank you for your answer Ken.

Kind regards,
Kuba


From sagar at transvoicesolutions.com  Mon Aug 13 14:56:13 2012
From: sagar at transvoicesolutions.com (Sagar K)
Date: Mon, 13 Aug 2012 16:26:13 +0530
Subject: [Freeswitch-users] connecting database in freeswitch using php
Message-ID: <CAKuapdOCkWvbtgD4a5PXOcoQvH-D2PJFFSSusZAPU3zBfCzYxQ@mail.gmail.com>

Hi, i'm new to this freeswitch,i'm not able to connect database in
freeswitch so pls guide me to how to configure the same..


Regards
Sagar K
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From info at pripojtese.net  Mon Aug 13 16:33:29 2012
From: info at pripojtese.net (Jakub Tencl)
Date: Mon, 13 Aug 2012 13:33:29 +0100
Subject: [Freeswitch-users] script for generating bulk calls
In-Reply-To: <CAKV-jFzWQ1U6f2f3zEP0mjdO3MvHW4f5u860q4LfoYhUZLARGA@mail.gmail.com>
References: <5025094E.4090802@healings.org.uk>
	<50250A09.5000409@pripojtese.net>
	<CAKV-jFzWQ1U6f2f3zEP0mjdO3MvHW4f5u860q4LfoYhUZLARGA@mail.gmail.com>
Message-ID: <5028F419.2080700@pripojtese.net>

The requests is from 10 to 50 concurent calls with specify call duration 
and to specify range of the numbers, if this is what do you have, kind 
of similar thing, can you afford it for some conditions please?

Thanks

Jakub Tencl

Dne 11/08/2012 02:43, Brian Foster napsal(a):
>
> Newfies Dialer or have it contracted out. We've done something similar 
> here.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
>
> On Aug 10, 2012 5:45 PM, "Jakub Tencl" <info at pripojtese.net 
> <mailto:info at pripojtese.net>> wrote:
>
>
>
>     Hello,
>
>     i am quite new with freeswitch. Can anybody tell me if Freeswitch has
>     module for generating bulk calls and if not can you afford some
>     template
>     script for it or just some pointers where i have to go?
>
>     Thank you
>
>     Best Regards
>
>     Jakub
>
>
>
>
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     FreeSWITCH-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
>
>
> -- 
> This message has been scanned for viruses and
> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is
> believed to be clean.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

Kind Regards

Jakub Tencl

*Mobil:* +44 770 4734 834 or +420 774 430 010| *ICQ:* 262823686| 
*Skype:* spycat108

This message and any attachment are confidential and may be privileged 
or  otherwise protected from disclosure. If you are not the intended 
recipient,please telephone or email the sender and delete this message 
and any attachment from your system. If you are not the intended 
recipient you  must not copy this message or attachment or disclose the 
contents to any other person.

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From info at pripojtese.net  Mon Aug 13 16:38:38 2012
From: info at pripojtese.net (Jakub Tencl)
Date: Mon, 13 Aug 2012 13:38:38 +0100
Subject: [Freeswitch-users] script for generating bulk calls
In-Reply-To: <CAAdTMa-NqDg6rk-7+9QPJ7ZGTsQJcmApyd72WzmgY6MCq1TrOw@mail.gmail.com>
References: <5025094E.4090802@healings.org.uk>
	<50250A09.5000409@pripojtese.net>
	<CAKV-jFzWQ1U6f2f3zEP0mjdO3MvHW4f5u860q4LfoYhUZLARGA@mail.gmail.com>
	<CAAdTMa-NqDg6rk-7+9QPJ7ZGTsQJcmApyd72WzmgY6MCq1TrOw@mail.gmail.com>
Message-ID: <5028F54E.20903@pripojtese.net>

I used to now vBilling open source system, i am afraid that another 
system like fusionpbx gonna make mess in original installation of the 
freeswitch which was installed belonging to vBilling so it's quite 
unusual installation for example folder is on the diferent place, not 
/usr/local/freeswitch

Dne 11/08/2012 08:41, BookBag napsal(a):
>
> I had this problem and ended up first creating a script of a whole 
> bunch originate statements. Then  I found fusionpbx. It has a call 
> broadcast script already built it using a web interface. You might 
> wanna take a look at it.
>
> On Aug 10, 2012 9:47 PM, "Brian Foster" <bdfoster at endigotech.com 
> <mailto:bdfoster at endigotech.com>> wrote:
>
>     Newfies Dialer or have it contracted out. We've done something
>     similar here.
>
>     Brian Foster
>     Endigo Computer LLC
>
>     Sent from a mobile device.
>
>     On Aug 10, 2012 5:45 PM, "Jakub Tencl" <info at pripojtese.net
>     <mailto:info at pripojtese.net>> wrote:
>
>
>
>         Hello,
>
>         i am quite new with freeswitch. Can anybody tell me if
>         Freeswitch has
>         module for generating bulk calls and if not can you afford
>         some template
>         script for it or just some pointers where i have to go?
>
>         Thank you
>
>         Best Regards
>
>         Jakub
>
>
>
>
>
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>
>         Join Us At ClueCon - Aug 7-9, 2012
>
>         FreeSWITCH-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
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     Join Us At ClueCon - Aug 7-9, 2012
>
>     FreeSWITCH-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
>
>
> -- 
> This message has been scanned for viruses and
> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is
> believed to be clean.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

Kind Regards

Jakub Tencl

*Mobil:* +44 770 4734 834 or +420 774 430 010| *ICQ:* 262823686| 
*Skype:* spycat108

This message and any attachment are confidential and may be privileged 
or  otherwise protected from disclosure. If you are not the intended 
recipient,please telephone or email the sender and delete this message 
and any attachment from your system. If you are not the intended 
recipient you  must not copy this message or attachment or disclose the 
contents to any other person.

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From th982a at googlemail.com  Mon Aug 13 17:57:01 2012
From: th982a at googlemail.com (Tamer Higazi)
Date: Mon, 13 Aug 2012 15:57:01 +0200
Subject: [Freeswitch-users] make fs listen to internal ip-adress
In-Reply-To: <CAMWiMy_bEnMJ1f=XFiZ=OPKm21tCaKBGnFRfT2oxrLvDB4k1HQ@mail.gmail.com>
References: <50282862.2090200@googlemail.com>
	<CAMWiMy_bEnMJ1f=XFiZ=OPKm21tCaKBGnFRfT2oxrLvDB4k1HQ@mail.gmail.com>
Message-ID: <502907AD.6030905@googlemail.com>

Solved!
My own stupidy. It works all fine.....

Am 13.08.2012 02:37, schrieb curriegrad2004:
> Are you sure there's another profile that isn't listening on to the
> same ports where your "internal" profile is trying to bind to?
> 
> On Sun, Aug 12, 2012 at 3:04 PM, Tamer Higazi <th982a at googlemail.com> wrote:
>> Hi people!
>> I have the problem making freeswitch listen to an internal IP-Adress,
>> that trunks in the internal network, connected at a specific NIC board,
>> are able to register.
>>
>> eth0 is connected to a DSL modem, eth3 my ATA Adapter, with 2 Trunks
>> that should register / login at freeswitch.
>>
>>
>> in the CLI, "sofia status" tells me, that "internal", is the WAN-IPadress:
>>
>> internal    profile             sip:mod_sofia at 212.255.33.187:5060
>> RUNNING (0)
>>
>>
>> what I of course don't like.
>>
>> if I set in the internal.xml,
>>
>> <param name="rtp-ip" value="192.168.2.10"/>
>> <param name="sip-ip" value="192.168.2.10"/>
>>
>> I get the error on the console:
>>
>> here is the error: 2012-08-12 23:42:12.107512 [ERR] sofia.c:2289 Error
>> Creating SIP UA for profile: internal
>>
>> How do I change the setting in "internal.xml" proparly, so that FS
>> listens to everything in 192.168.2.10....
>>
>>
>>
>> I would kindly thank you
>>
>>
>>
>> Tamer
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From asaad2 at gmail.com  Mon Aug 13 18:50:05 2012
From: asaad2 at gmail.com (BookBag)
Date: Mon, 13 Aug 2012 09:50:05 -0500
Subject: [Freeswitch-users] script for generating bulk calls
In-Reply-To: <5028F419.2080700@pripojtese.net>
References: <5025094E.4090802@healings.org.uk>
	<50250A09.5000409@pripojtese.net>
	<CAKV-jFzWQ1U6f2f3zEP0mjdO3MvHW4f5u860q4LfoYhUZLARGA@mail.gmail.com>
	<5028F419.2080700@pripojtese.net>
Message-ID: <CAAdTMa_NOie9jpsxN4L+4+G1ik=cjjMr-91M1HX_8m3KXm9CVw@mail.gmail.com>

no, mine was a set amount of numbers that were already know and they played
a wav file, also my provider only handled 2 concurrent channels at a time.
so it would dial 2 numbers at time. Thats why fusionpbx was such a help
because it allowed me to do that and all I had to do was cut and paste the
numbers I was going to call.

On Mon, Aug 13, 2012 at 7:33 AM, Jakub Tencl <info at pripojtese.net> wrote:

>  The requests is from 10 to 50 concurent calls with specify call duration
> and to specify range of the numbers, if this is what do you have, kind of
> similar thing, can you afford it for some conditions please?
>
> Thanks
>
> Jakub Tencl
>
> Dne 11/08/2012 02:43, Brian Foster napsal(a):
>
> Newfies Dialer or have it contracted out. We've done something similar
> here.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Aug 10, 2012 5:45 PM, "Jakub Tencl" <info at pripojtese.net> wrote:
>
>>
>>
>> Hello,
>>
>> i am quite new with freeswitch. Can anybody tell me if Freeswitch has
>> module for generating bulk calls and if not can you afford some template
>> script for it or just some pointers where i have to go?
>>
>> Thank you
>>
>> Best Regards
>>
>> Jakub
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> --
> This message has been scanned for viruses and
> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is
> believed to be clean.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>
>
> --
>
> Kind Regards
>
> Jakub Tencl
>
> *Mobil:* +44 770 4734 834 or +420 774 430 010| *ICQ:* 262823686 | *Skype:*spycat108
>
> This message and any attachment are confidential and may be privileged or
> otherwise protected from disclosure. If you are not the intended
> recipient,please telephone or email the sender and delete this message and
> any  attachment from your system. If you are not the intended recipient
> you  must not copy this message or attachment or disclose the contents to
> any other  person.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From admin at blindi.net  Mon Aug 13 18:53:10 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Mon, 13 Aug 2012 16:53:10 +0200 (CEST)
Subject: [Freeswitch-users] fs crashs after more then 100 dtmf events on
 a call
In-Reply-To: <1FFF97C269757C458224B7C895F35F1514327E@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1514327E@cantor.std.visionutv.se>
Message-ID: <Pine.LNX.4.64.1208131621500.19780@1a-8140.antagus.de>

Hi peter, thanks for you helpful message.

I describe a easy way to crash fs.

  You can reproduce this situation.

step1
recording 3 soundfiles and store in a directory.

step2
the luacode:

-- simply script to crash freeswitch
--
session:sleep(100);
session:answer()
session:sleep(200);
  session:setInputCallback("onInput", "");
function onInput(s, type, obj)
   freeswitch.consoleLog("info", "Callback with type " .. type .. "\n");
--  freeswitch.consoleLog("info", "DTMF Digit: " .. obj.digit .. "\n");

   if ( obj['digit'] == '1' ) then
-- if (obj.digit   == "1")  then 
--  session:streamFile(msg_file_play)
session:streamFile("/usr/local/freeswitch/sounds/nummer1.alaw")
read_msg_menu()
  return("false");

end

   if ( obj['digit'] == '2' ) then
-- if (obj.digit   == "2")  then 
--  session:streamFile(msg_file_play)
session:streamFile("/usr/local/freeswitch/sounds/nummer2.alaw")
read_msg_menu()
  return("false");

end

   if ( obj['digit'] == '3' ) then
-- if (obj.digit   == "3")  then 
--  session:streamFile(msg_file_play)
session:streamFile("/usr/local/freeswitch/sounds/nummer3.alaw")
read_msg_menu()
  return("false");

end

end

function read_msg_menu( )
  session:setInputCallback("onInput");
session:streamFile("/usr/local/freeswitch/sounds/nummer1.alaw")
end


  read_msg_menu()

step3
put this in your dialplan.
for example:
<!-- crash fs via simply  luascript  -->
<extension name="crashtest" >
<condition field="destination_number" expression="^104$">
<action application="lua" data="simply-crash.lua"/>
</condition>
</extension>

step4 ring the extension.
Heare the testprompt.
Press 60  70  times 1 during the announcement.
fs crashs.

The system is from current git:
FreeSWITCH Version 1.2.0-rc2+git~20120813T041310Z~aad07c6243 (1.2.0-rc2; 
git at
commit aad07c6243 on Mon, 13 Aug 2012 04:13:10 Z)

This is very danger, then to plan a phonecast (newsystem) or public
voiceforums.
thanks.


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From asaad2 at gmail.com  Mon Aug 13 18:54:31 2012
From: asaad2 at gmail.com (BookBag)
Date: Mon, 13 Aug 2012 09:54:31 -0500
Subject: [Freeswitch-users] script for generating bulk calls
In-Reply-To: <CAAdTMa_NOie9jpsxN4L+4+G1ik=cjjMr-91M1HX_8m3KXm9CVw@mail.gmail.com>
References: <5025094E.4090802@healings.org.uk>
	<50250A09.5000409@pripojtese.net>
	<CAKV-jFzWQ1U6f2f3zEP0mjdO3MvHW4f5u860q4LfoYhUZLARGA@mail.gmail.com>
	<5028F419.2080700@pripojtese.net>
	<CAAdTMa_NOie9jpsxN4L+4+G1ik=cjjMr-91M1HX_8m3KXm9CVw@mail.gmail.com>
Message-ID: <CAAdTMa8WnCEs8v_ouckjVHb5BBWX2z=kHAPpswCCZXrejsGu0Q@mail.gmail.com>

yeah I see your dillemma, I dont know if vbilling allows you to get to
fs_cli , if if it does then you can just enter a whole bunch of originate
statements. you can probably use excel to create a massive csv file then
copy and paste that into fs_cli which will then call the range of phone
numbers you need.

On Mon, Aug 13, 2012 at 9:50 AM, BookBag <asaad2 at gmail.com> wrote:

> no, mine was a set amount of numbers that were already know and they
> played a wav file, also my provider only handled 2 concurrent channels at a
> time. so it would dial 2 numbers at time. Thats why fusionpbx was such a
> help because it allowed me to do that and all I had to do was cut and paste
> the numbers I was going to call.
>
>
> On Mon, Aug 13, 2012 at 7:33 AM, Jakub Tencl <info at pripojtese.net> wrote:
>
>>  The requests is from 10 to 50 concurent calls with specify call
>> duration and to specify range of the numbers, if this is what do you have,
>> kind of similar thing, can you afford it for some conditions please?
>>
>> Thanks
>>
>> Jakub Tencl
>>
>> Dne 11/08/2012 02:43, Brian Foster napsal(a):
>>
>> Newfies Dialer or have it contracted out. We've done something similar
>> here.
>>
>> Brian Foster
>> Endigo Computer LLC
>>
>> Sent from a mobile device.
>> On Aug 10, 2012 5:45 PM, "Jakub Tencl" <info at pripojtese.net> wrote:
>>
>>>
>>>
>>> Hello,
>>>
>>> i am quite new with freeswitch. Can anybody tell me if Freeswitch has
>>> module for generating bulk calls and if not can you afford some template
>>> script for it or just some pointers where i have to go?
>>>
>>> Thank you
>>>
>>> Best Regards
>>>
>>> Jakub
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> --
>> This message has been scanned for viruses and
>> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and
>> is
>> believed to be clean.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> 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
>>
>>
>>
>> --
>>
>> Kind Regards
>>
>> Jakub Tencl
>>
>> *Mobil:* +44 770 4734 834 or +420 774 430 010| *ICQ:* 262823686 | *Skype:
>> * spycat108
>>
>> This message and any attachment are confidential and may be privileged
>> or  otherwise protected from disclosure. If you are not the intended
>> recipient,please telephone or email the sender and delete this message and
>> any  attachment from your system. If you are not the intended recipient
>> you  must not copy this message or attachment or disclose the contents to
>> any other  person.
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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 Aug 13 19:11:01 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 13 Aug 2012 17:11:01 +0200
Subject: [Freeswitch-users] script for generating bulk calls
In-Reply-To: <CAAdTMa8WnCEs8v_ouckjVHb5BBWX2z=kHAPpswCCZXrejsGu0Q@mail.gmail.com>
References: <5025094E.4090802@healings.org.uk>
	<50250A09.5000409@pripojtese.net>
	<CAKV-jFzWQ1U6f2f3zEP0mjdO3MvHW4f5u860q4LfoYhUZLARGA@mail.gmail.com>
	<5028F419.2080700@pripojtese.net>
	<CAAdTMa_NOie9jpsxN4L+4+G1ik=cjjMr-91M1HX_8m3KXm9CVw@mail.gmail.com>
	<CAAdTMa8WnCEs8v_ouckjVHb5BBWX2z=kHAPpswCCZXrejsGu0Q@mail.gmail.com>
Message-ID: <CAFGRPVo0aLFfBEbABFuAMVu2W0ZXiXT4_ue7D10pbUG2KVCOfA@mail.gmail.com>

Maybe you can use this one as a start:

It uses a MySQL table (below) to dial out numbers and parking them for the
time specified in the table.
All you need to do is originate in background not waiting for the result...


#!/usr/bin/perl

use DBI;
require ESL;


MysqlConnect();

my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
my $mydomain = "myserverIP"

my $sql_numbers = "select * from test_numbers where enabled = 1;";
my $sth_numbers = $db->prepare($sql_numbers);

if (!$sth_numbers->execute()){
     print "Could NOT execute!\n$sql_numbers\n";
}else{
     if(($rows_numbers = $sth_numbers->rows)>0){
          while (my $ref_numbers = $sth_numbers->fetchrow_hashref() ) {

               my $dialstr = "{execute_on_answer='sched_hangup +" .
$$ref_numbers{'len_park'} . " alloted_timeout',originate_retries=" .
$$ref_numbers{'tries'} . ",originate_retry_sleep_ms=1000}sofia/internal/" .
$$ref_numbers{'number'} . "%" . $mydomain . " &park()";
               print "DIALSTR: $dialstr \n";

               my $e = $con->api("originate", $dialstr);
               print $e->getBody();
               my $uuid = $e->getBody();

               SetVar($uuid,"customer_company","DIALER");
          }
     }
}

print "\n";
exit 0;


sub SetVar($$$){
     my $uuid = shift;
     my $var = shift;
     my $val = shift;
     do_api("uuid_setvar", $uuid . " " . $var . " " . $val);
     return;
}



sub do_api($$) {
     my $cmd = shift;
     my $args = shift;
     print "CMD : <$cmd>\nARGS: <$args>\n";
     my $e = $con->api($cmd, $args);
     if ($e) {
          print STDERR $e->getBody() . "\n";
     }
     return;
}

sub MysqlConnect(){
        $db = DBI->connect('DBI:mysql:dialer:127.0.0.1','root','password');
        $db->{mysql_auto_reconnect} = 1;
        if ($db<=0){
            print "Error connecting to data server!!!\n" if $debug;
            exit;
        }
}

sub fsLog($){
     my $msg = shift;
     $con->api("log", "WARNING " . $msg);
}



################################################################################
#
#          CREATE TABLE `test_numbers` (
#               `id` int(11) NOT NULL auto_increment,
#               `number` varchar(25) NOT NULL,
#               `tries` int(11) NOT NULL,
#               `len_park` int(11) NOT NULL,
#               `enabled` int(11) NOT NULL,
#               PRIMARY KEY  (`id`)
#          ) ENGINE=MyISAM AUTO_INCREMENT=2 DEFAULT CHARSET=utf8
#
################################################################################



Good luck.

David

On Mon, Aug 13, 2012 at 4:54 PM, BookBag <asaad2 at gmail.com> wrote:

> yeah I see your dillemma, I dont know if vbilling allows you to get to
> fs_cli , if if it does then you can just enter a whole bunch of originate
> statements. you can probably use excel to create a massive csv file then
> copy and paste that into fs_cli which will then call the range of phone
> numbers you need.
>
>
> On Mon, Aug 13, 2012 at 9:50 AM, BookBag <asaad2 at gmail.com> wrote:
>
>> no, mine was a set amount of numbers that were already know and they
>> played a wav file, also my provider only handled 2 concurrent channels at a
>> time. so it would dial 2 numbers at time. Thats why fusionpbx was such a
>> help because it allowed me to do that and all I had to do was cut and paste
>> the numbers I was going to call.
>>
>>
>> On Mon, Aug 13, 2012 at 7:33 AM, Jakub Tencl <info at pripojtese.net> wrote:
>>
>>>  The requests is from 10 to 50 concurent calls with specify call
>>> duration and to specify range of the numbers, if this is what do you have,
>>> kind of similar thing, can you afford it for some conditions please?
>>>
>>> Thanks
>>>
>>> Jakub Tencl
>>>
>>> Dne 11/08/2012 02:43, Brian Foster napsal(a):
>>>
>>> Newfies Dialer or have it contracted out. We've done something similar
>>> here.
>>>
>>> Brian Foster
>>> Endigo Computer LLC
>>>
>>> Sent from a mobile device.
>>> On Aug 10, 2012 5:45 PM, "Jakub Tencl" <info at pripojtese.net> wrote:
>>>
>>>>
>>>>
>>>> Hello,
>>>>
>>>> i am quite new with freeswitch. Can anybody tell me if Freeswitch has
>>>> module for generating bulk calls and if not can you afford some template
>>>> script for it or just some pointers where i have to go?
>>>>
>>>> Thank you
>>>>
>>>> Best Regards
>>>>
>>>> Jakub
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> --
>>> This message has been scanned for viruses and
>>> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and
>>> is
>>> believed to be clean.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> 
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> 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
>>>
>>>
>>>
>>> --
>>>
>>> Kind Regards
>>>
>>> Jakub Tencl
>>>
>>> *Mobil:* +44 770 4734 834 or +420 774 430 010| *ICQ:* 262823686 | *
>>> Skype:* spycat108
>>>
>>> This message and any attachment are confidential and may be privileged
>>> or  otherwise protected from disclosure. If you are not the intended
>>> recipient,please telephone or email the sender and delete this message and
>>> any  attachment from your system. If you are not the intended recipient
>>> you  must not copy this message or attachment or disclose the contents to
>>> any other  person.
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From danb.lists at gmail.com  Mon Aug 13 19:15:07 2012
From: danb.lists at gmail.com (DanB)
Date: Mon, 13 Aug 2012 17:15:07 +0200
Subject: [Freeswitch-users] Status mod_celliax
In-Reply-To: <mailman.16688.1344869743.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.16688.1344869743.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <502919FB.5020700@gmail.com>

Hey Guys,

Since I could not find a clear spot in the wiki regarding mod_celliax, I 
was wondering if anyone could help me with some information regarding 
it's status. Did it ever end into FreeSWITCH or was just planned to be 
and no interest of any party? I have seen quite some time ago some 
discussions regarding integrating it into FS as well hence my questions.

Thanks in advance!

DanB


From peter.olsson at visionutveckling.se  Mon Aug 13 19:36:41 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 13 Aug 2012 15:36:41 +0000
Subject: [Freeswitch-users] fs crashs after more then 100 dtmf events on
 a call
In-Reply-To: <Pine.LNX.4.64.1208131621500.19780@1a-8140.antagus.de>
References: <1FFF97C269757C458224B7C895F35F1514327E@cantor.std.visionutv.se>,
	<Pine.LNX.4.64.1208131621500.19780@1a-8140.antagus.de>
Message-ID: <E19ECE61-5F21-48B6-8722-D510B05A74C0@visionutveckling.se>

Please report to Jira, or it will probably just get lost somewhere on this list.

/Peter

13 aug 2012 kl. 16:58 skrev "Thomas Hoellriegel" <admin at blindi.net>:

> Hi peter, thanks for you helpful message.
> 
> I describe a easy way to crash fs.
> 
> You can reproduce this situation.
> 
> step1
> recording 3 soundfiles and store in a directory.
> 
> step2
> the luacode:
> 
> -- simply script to crash freeswitch
> --
> session:sleep(100);
> session:answer()
> session:sleep(200);
> session:setInputCallback("onInput", "");
> function onInput(s, type, obj)
>  freeswitch.consoleLog("info", "Callback with type " .. type .. "\n");
> --  freeswitch.consoleLog("info", "DTMF Digit: " .. obj.digit .. "\n");
> 
>  if ( obj['digit'] == '1' ) then
> -- if (obj.digit   == "1")  then --  session:streamFile(msg_file_play)
> session:streamFile("/usr/local/freeswitch/sounds/nummer1.alaw")
> read_msg_menu()
> return("false");
> 
> end
> 
>  if ( obj['digit'] == '2' ) then
> -- if (obj.digit   == "2")  then --  session:streamFile(msg_file_play)
> session:streamFile("/usr/local/freeswitch/sounds/nummer2.alaw")
> read_msg_menu()
> return("false");
> 
> end
> 
>  if ( obj['digit'] == '3' ) then
> -- if (obj.digit   == "3")  then --  session:streamFile(msg_file_play)
> session:streamFile("/usr/local/freeswitch/sounds/nummer3.alaw")
> read_msg_menu()
> return("false");
> 
> end
> 
> end
> 
> function read_msg_menu( )
> session:setInputCallback("onInput");
> session:streamFile("/usr/local/freeswitch/sounds/nummer1.alaw")
> end
> 
> 
> read_msg_menu()
> 
> step3
> put this in your dialplan.
> for example:
> <!-- crash fs via simply  luascript  -->
> <extension name="crashtest" >
> <condition field="destination_number" expression="^104$">
> <action application="lua" data="simply-crash.lua"/>
> </condition>
> </extension>
> 
> step4 ring the extension.
> Heare the testprompt.
> Press 60  70  times 1 during the announcement.
> fs crashs.
> 
> The system is from current git:
> FreeSWITCH Version 1.2.0-rc2+git~20120813T041310Z~aad07c6243 (1.2.0-rc2; git at
> commit aad07c6243 on Mon, 13 Aug 2012 04:13:10 Z)
> 
> This is very danger, then to plan a phonecast (newsystem) or public
> voiceforums.
> thanks.
> 
> 
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
> 
> 
> !DSPAM:5029127b32767892227582!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:5029127b32767892227582!


From vipkilla at gmail.com  Mon Aug 13 19:50:25 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 13 Aug 2012 11:50:25 -0400
Subject: [Freeswitch-users] connecting database in freeswitch using php
In-Reply-To: <CAKuapdOCkWvbtgD4a5PXOcoQvH-D2PJFFSSusZAPU3zBfCzYxQ@mail.gmail.com>
References: <CAKuapdOCkWvbtgD4a5PXOcoQvH-D2PJFFSSusZAPU3zBfCzYxQ@mail.gmail.com>
Message-ID: <CAC-LwPPWbEwn3nxLDN6hKcDsD65BkKqR40o2UyrhYwg4TTU5Jg@mail.gmail.com>

Write a php script that generates XML based on the database and use
mod_xml_curl to parse the generated XML

On Mon, Aug 13, 2012 at 6:56 AM, Sagar K <sagar at transvoicesolutions.com> wrote:
> Hi, i'm new to this freeswitch,i'm not able to connect database in
> freeswitch so pls guide me to how to configure the same..
>
>
> Regards
> Sagar K
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From vipkilla at gmail.com  Mon Aug 13 19:52:10 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 13 Aug 2012 11:52:10 -0400
Subject: [Freeswitch-users] The domain_name variable and domains in
	FreeSWITCH
In-Reply-To: <20120813090751.GA15900@willow.kavun.ch>
References: <20120813090751.GA15900@willow.kavun.ch>
Message-ID: <CAC-LwPPhp=d6w1XkAx_1pj1=J6JbjWRF9Yvx1Sno+GKC1KkPQQ@mail.gmail.com>

If properly configured you can have each domain operate as a different
tenant on the server, so they will be separate PBXs.

On Mon, Aug 13, 2012 at 5:07 AM, Emrah <eka at ekanet.net> wrote:
> Hello all,
>
> I have been an Asterisk user for almost 10 years... God I'm glad I took the leap and tried FS!
> Congratulations to all of you for the amazing dev you've put together.
>
> I have been trying to understand in depth how the domain_name variable is used throughout FS. Or more in general, what defines a domain name and how is it used?
>
> It doesn't seem that FS is domain sensitive for registrations / authenticated calls. If both sip.example.com and fs.example.com are IN A 1.2.3.4, I can register on both domains with no
> distinction whatsoever.
>
> How does that work?
>
> Best to you and thanks for your precious answers.
> --
> Emrah
> <eka at ekanet.net>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From gmaruzz at gmail.com  Mon Aug 13 20:24:39 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 13 Aug 2012 11:24:39 -0500
Subject: [Freeswitch-users] Status mod_celliax
In-Reply-To: <502919FB.5020700@gmail.com>
References: <mailman.16688.1344869743.13443.freeswitch-users@lists.freeswitch.org>
	<502919FB.5020700@gmail.com>
Message-ID: <CALXCt0o6o2S_sT9RzyJuDkWWg7gouucGEDR_iSb3Y2uc2GWqKw@mail.gmail.com>

it's calle mod-gsmopen now, and it's part of FreeSWITCH.
Look for it in the FS wiki: http://wiki.freeswitch.org
-giovanni

On 8/13/12, DanB <danb.lists at gmail.com> wrote:
> Hey Guys,
>
> Since I could not find a clear spot in the wiki regarding mod_celliax, I
> was wondering if anyone could help me with some information regarding
> it's status. Did it ever end into FreeSWITCH or was just planned to be
> and no interest of any party? I have seen quite some time ago some
> discussions regarding integrating it into FS as well hence my questions.
>
> Thanks in advance!
>
> DanB
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 leonardo at daitangroup.com  Mon Aug 13 21:29:25 2012
From: leonardo at daitangroup.com (Leonardo)
Date: Mon, 13 Aug 2012 14:29:25 -0300
Subject: [Freeswitch-users] can I set the "api originate" command to use
	dialplan?
Message-ID: <50293975.9090401@daitangroup.com>

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From hexade at hotmail.com  Mon Aug 13 22:53:22 2012
From: hexade at hotmail.com (Adelia C.)
Date: Mon, 13 Aug 2012 14:53:22 -0400
Subject: [Freeswitch-users] Use of the Call Recording LAME library (.wav ->
 .mp3) - free or licensed?
Message-ID: <BLU002-W227ABE50A0D931D8C29D06DDAB00@phx.gbl>

For those who use FreeSwitch for commercial purposes in the US - did you have to buy any license to use LAME? 
Need to translate from .wav to .mo3 to play the recording. Do I need to worry? WinLAME README:



?Note that personal and/or
commercial use of compiled versions of the LAME encoding engine (including the
DLL distributed with winLAME) requires a   patent license in some
countries. Check before using winLAME!?



Thanks,
A.C.
 		 	   		  
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From amilkhanzada at gmail.com  Mon Aug 13 23:42:05 2012
From: amilkhanzada at gmail.com (Amil)
Date: Mon, 13 Aug 2012 12:42:05 -0700
Subject: [Freeswitch-users] Reloading mod_python dist-packages?
In-Reply-To: <CAOTFBbeK8O4Sa3jQfLjcamngG_mM6qg4yxH8uLc3rsdPGZYobw@mail.gmail.com>
References: <CAOTFBbcmUKiw+o+t0OzRJg8atQ8mh5S2jP+5hbTDcq=aAwViBw@mail.gmail.com>
	<CAOTFBbeK8O4Sa3jQfLjcamngG_mM6qg4yxH8uLc3rsdPGZYobw@mail.gmail.com>
Message-ID: <CAOTFBbexYfRknDm2Hi2K+v9MOGjPbZjUBiqorYpGHc+9-M-tVQ@mail.gmail.com>

*last bump*

On Thu, Jul 19, 2012 at 12:17 PM, Amil <amilkhanzada at gmail.com> wrote:

> Any ideas? If it is not possible, please let me know.
>
> Thanks,
> Amil
>
> http://www.amilkhanzada.com/2012/06/gmail-keeping-mailing-list-topics-you.html
>
>
> On Sun, Jul 15, 2012 at 8:57 PM, Amil <amilkhanzada at gmail.com> wrote:
>
>> So, I am writing some python scripts that are copied into the
>> dist-packages directory of python.
>> However, I am also using python scripts in the FreeSWITCH scripts
>> directory that refer to these python scripts in the dist-packages.
>> To re-copy the files, I need to run "sudo python setup_fs.py" each time I
>> make a change.
>>
>> However, I also need to restart FreeSWITCH because I don't know of a way
>> to reload the dist-packages used by mod_python.
>>
>> Any ideas?
>>
>> PS: This is the project I am referring to: <
>> https://github.com/kheimerl/libvbts>
>>
>
>
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From roger.castaldo at gmail.com  Tue Aug 14 00:27:56 2012
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Mon, 13 Aug 2012 16:27:56 -0400
Subject: [Freeswitch-users] Pinned Routing in Dialplan
Message-ID: <CAAYf2LsMxzCJ5DWBAhw+wSLiroQLOr-x-hjZ_KAMuQ=BEwfCmw@mail.gmail.com>

Hi everyone I am trying to setup some pinned routing in freeswitch strictly
through the xml dialplan, so not using any external scripts to validate
pins, that being said i have written a sample dial plan below and it seems
to fail, not sure why.

The parts that are failing are actually detecting a valid pin was placed in
as well as hanging up the phone call when more than 3 attempts have been
made.  can anyone please help me figure out why this is not working.

<extension name="pinned_route_test" inline="true">
      <condition field="destination_number" expression="^5555$">
        <action application="answer"/>
        <action application="log" data="INFO Attempting to obtain pin"/>
        <action application="play_and_get_digits" data="4 10 3 3000 #
conference\8000\conf-pin.wav silence_stream://250 inputted_validation_pin
\d+"/>
        <action application="log" data="INFO Pin Number obtained
${inputted_validation_pin}."/>
        <action application="set"
data="pin_attempts=${expr(${pin_attempts}+1)}"/>
        <action application="log" data="INFO # of pin attempts
${pin_attempts}."/>
      </condition>
      <condition field="inputted_validation_pin" expression="1234"
break="on-true">
        <action application="log" data="INFO Pin number entered was valid"/>
        <anti-action application="log" data="INFO 1234 is not a valid match
for the inputted pin ${inputted_validation_pin}."/>
      </condition>
      <condition field="pin_attempts" expression="3">
        <action application="playback"
data="conference\8000\conf-goodbye.wav"/>
        <action application="hangup"/>
        <anti-action application="log" data="INFO invalid pin was inputted
for attemp ${pin_attempts}."/>
        <anti-action application="playback"
data="conference\8000\conf-bad-pin.wav"/>
        <anti-action application="log" data="INFO restarting extension
pinned_route_test."/>
        <anti-action application="transfer" data="${destination_number} XML
default"/>
      </condition>
    </extension>
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From jmesquita at freeswitch.org  Tue Aug 14 01:27:12 2012
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 13 Aug 2012 18:27:12 -0300
Subject: [Freeswitch-users] Reloading mod_python dist-packages?
In-Reply-To: <CAOTFBbexYfRknDm2Hi2K+v9MOGjPbZjUBiqorYpGHc+9-M-tVQ@mail.gmail.com>
References: <CAOTFBbcmUKiw+o+t0OzRJg8atQ8mh5S2jP+5hbTDcq=aAwViBw@mail.gmail.com>
	<CAOTFBbeK8O4Sa3jQfLjcamngG_mM6qg4yxH8uLc3rsdPGZYobw@mail.gmail.com>
	<CAOTFBbexYfRknDm2Hi2K+v9MOGjPbZjUBiqorYpGHc+9-M-tVQ@mail.gmail.com>
Message-ID: <CABH8QX3NQPTMS3U-=JfOrV5Fwn5ekU0tEdRyYAsdtRrOmYZsWA@mail.gmail.com>

Reloading the mod should be enough. I imagine that mod_python respects
anything that the interpreter would do as if it was running standalone.

Reloading modules is potentially hazardous and you should be careful with
it:
http://stackoverflow.com/questions/684171/how-to-re-import-an-updated-package-while-in-python-interpreter

Regards,
Jo?o Mesquita



On Mon, Aug 13, 2012 at 4:42 PM, Amil <amilkhanzada at gmail.com> wrote:

> *last bump*
>
>
> On Thu, Jul 19, 2012 at 12:17 PM, Amil <amilkhanzada at gmail.com> wrote:
>
>> Any ideas? If it is not possible, please let me know.
>>
>> Thanks,
>> Amil
>>
>> http://www.amilkhanzada.com/2012/06/gmail-keeping-mailing-list-topics-you.html
>>
>>
>> On Sun, Jul 15, 2012 at 8:57 PM, Amil <amilkhanzada at gmail.com> wrote:
>>
>>> So, I am writing some python scripts that are copied into the
>>> dist-packages directory of python.
>>> However, I am also using python scripts in the FreeSWITCH scripts
>>> directory that refer to these python scripts in the dist-packages.
>>> To re-copy the files, I need to run "sudo python setup_fs.py" each time
>>> I make a change.
>>>
>>> However, I also need to restart FreeSWITCH because I don't know of a way
>>> to reload the dist-packages used by mod_python.
>>>
>>> Any ideas?
>>>
>>> PS: This is the project I am referring to: <
>>> https://github.com/kheimerl/libvbts>
>>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From amilkhanzada at gmail.com  Tue Aug 14 01:47:22 2012
From: amilkhanzada at gmail.com (Amil)
Date: Mon, 13 Aug 2012 14:47:22 -0700
Subject: [Freeswitch-users] Reloading mod_python dist-packages?
In-Reply-To: <CABH8QX3NQPTMS3U-=JfOrV5Fwn5ekU0tEdRyYAsdtRrOmYZsWA@mail.gmail.com>
References: <CAOTFBbcmUKiw+o+t0OzRJg8atQ8mh5S2jP+5hbTDcq=aAwViBw@mail.gmail.com>
	<CAOTFBbeK8O4Sa3jQfLjcamngG_mM6qg4yxH8uLc3rsdPGZYobw@mail.gmail.com>
	<CAOTFBbexYfRknDm2Hi2K+v9MOGjPbZjUBiqorYpGHc+9-M-tVQ@mail.gmail.com>
	<CABH8QX3NQPTMS3U-=JfOrV5Fwn5ekU0tEdRyYAsdtRrOmYZsWA@mail.gmail.com>
Message-ID: <CAOTFBbcoYvvSvMzefvxUaNdG6_ROs3CidOJnGBZCazMFKkQMxQ@mail.gmail.com>

Hi Jo?o,

Thank you for the response. However, mod_python is not reloadable on my
machine:

> freeswitch at amil-ubuntu> reload mod_python
> 2012-08-13 13:44:48.922331 [CRIT] switch_loadable_module.c:1421 Module is
> not unloadable.
> -ERR unloading module [Module is not unloadable]
> +OK Reloading XML
> -ERR loading module [Module already loaded]
> 2012-08-13 13:44:49.242317 [WARNING] switch_loadable_module.c:1370 Module
> mod_python Already Loaded!
> 2012-08-13 13:44:49.242317 [INFO] mod_enum.c:812 ENUM Reloaded
> 2012-08-13 13:44:49.242317 [INFO] switch_time.c:1123 Timezone reloaded 530
> definitions



Any other ideas? Is there some command to just reload the dist-packages? If
not, I guess this is a feature request (although not urgent).

-Amil

On Mon, Aug 13, 2012 at 2:27 PM, Jo?o Mesquita <jmesquita at freeswitch.org>wrote:

> Reloading the mod should be enough. I imagine that mod_python respects
> anything that the interpreter would do as if it was running standalone.
>
> Reloading modules is potentially hazardous and you should be careful with
> it:
> http://stackoverflow.com/questions/684171/how-to-re-import-an-updated-package-while-in-python-interpreter
>
> Regards,
> Jo?o Mesquita
>
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From admin at blindi.net  Tue Aug 14 02:53:52 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 14 Aug 2012 00:53:52 +0200 (CEST)
Subject: [Freeswitch-users] fs crashs after more then 100 dtmf events on
 a call
In-Reply-To: <E19ECE61-5F21-48B6-8722-D510B05A74C0@visionutveckling.se>
References: <1FFF97C269757C458224B7C895F35F1514327E@cantor.std.visionutv.se>, 
	<Pine.LNX.4.64.1208131621500.19780@1a-8140.antagus.de>
	<E19ECE61-5F21-48B6-8722-D510B05A74C0@visionutveckling.se>
Message-ID: <Pine.LNX.4.64.1208140044060.21902@1a-8140.antagus.de>

Hi Peter,
sorry, im.m a blind user. I can only use a textbased browser lynx on 
linux.
The pastebin interface can.t be  used with text only browsers.

I have store the cord dump on my website:
You can download at:
www.blindi.net/core.gz
wget wget www.blindi.net/core.gz


fs breaks with a segfault.

Thanks.


---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From bdfoster at endigotech.com  Tue Aug 14 03:04:53 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 13 Aug 2012 19:04:53 -0400
Subject: [Freeswitch-users] can I set the "api originate" command to use
	dialplan?
In-Reply-To: <50293975.9090401@daitangroup.com>
References: <50293975.9090401@daitangroup.com>
Message-ID: <CAKV-jFx=TVu3rxk8N49P_DQ1CSTM9cmXNuoxV-bMsfO1anw2eQ@mail.gmail.com>

that's up to your script to provide. You'really already at the lowest level
to make a call via event socket.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 13, 2012 1:33 PM, "Leonardo" <leonardo at daitangroup.com> wrote:

>  Hi.
> I'm using the command "originate" to make a call via socket interface:
>
> api originate sofia/external/MY_PSTN_NUMBER at PSTN_GATEWAY &park()
> api originate sofia/external/MY_PSTN_NUMBER at PSTN_GATEWAY&bridge(originate sofia/external/OTHER_PSTN_NUMBER@
> PSTN_GATEWAY)
>
> Is there a way to omit the "PSTN_GATEWAY" and force the Freeswitch figure
> out which one to use based on PSTN_NUMBER/dialplan?
>
> Thanks i advance,
> Leo
>
>  --
>
> Leonardo N. S. Pereira, *Software Engineer*
> T:+55.19.3112-1200 ext. 1283*|*F:+55.19.3207-1437
> *Daitan**Group**|**www.daitangroup.com* <http://www.daitangroup.com/>*|**Highly
> Reliable Outsourcing. Value Added Services Worldwide.*
> *Privileged and confidential. If this message has been received in error,
> please notify sender and delete it immediately.*
> *Conte?do confidencial. Se esta mensagem foi recebida por engano, favor
> avisar o remetente e apag?-la imediatamente.*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Tue Aug 14 05:37:11 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 13 Aug 2012 21:37:11 -0400
Subject: [Freeswitch-users] The domain_name variable and domains in
	FreeSWITCH
In-Reply-To: <20120813090751.GA15900@willow.kavun.ch>
References: <20120813090751.GA15900@willow.kavun.ch>
Message-ID: <CAKV-jFzsCXq9S69cetFbgAvb=yWswUNXyk+O1YdYs52aXydt3Q@mail.gmail.com>

This falls into the multi tenant catagory. Please do a search on the wiki
for multitenant there are guides on how to change that behavior. Its a
relatively simple process. However you need to change certain aspects like
file locations to accomidate multiple tenants. By default $$domain is
generated as the local ipv4 address.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 13, 2012 9:43 AM, "Emrah" <eka at ekanet.net> wrote:

> Hello all,
>
> I have been an Asterisk user for almost 10 years... God I'm glad I took
> the leap and tried FS!
> Congratulations to all of you for the amazing dev you've put together.
>
> I have been trying to understand in depth how the domain_name variable is
> used throughout FS. Or more in general, what defines a domain name and how
> is it used?
>
> It doesn't seem that FS is domain sensitive for registrations /
> authenticated calls. If both sip.example.com and fs.example.com are IN A
> 1.2.3.4, I can register on both domains with no
> distinction whatsoever.
>
> How does that work?
>
> Best to you and thanks for your precious answers.
> --
> Emrah
> <eka at ekanet.net>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From manavid at gmail.com  Tue Aug 14 05:53:22 2012
From: manavid at gmail.com (Mohammad Amin Navid)
Date: Mon, 13 Aug 2012 18:53:22 -0700
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAKV-jFzmZQpUw6wWbsaDe-3Z0Q0J_0r3+fQV7dCbf_tg7hM3aA@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F15141094@cantor.std.visionutv.se>
	<50236299.4070403@softnet.si>
	<CAKV-jFyJEGW72UwZ8AGkKPL9uvoab3fOUb7y5_wTx6_suPT0JQ@mail.gmail.com>
	<CAKbxfG8Wavpt7COjPxm9BGA7LV-8_NZBS7koakMNTiTUBE-4=A@mail.gmail.com>
	<CAKV-jFzmZQpUw6wWbsaDe-3Z0Q0J_0r3+fQV7dCbf_tg7hM3aA@mail.gmail.com>
Message-ID: <09D78D0A-9F2F-41C1-B78D-B739D5D74B31@gmail.com>

I use fpm https://github.com/jordansissel/fpm/ for building packages, it's just awesome!

On Aug 11, 2012, at 5:50 PM, Brian Foster <bdfoster at endigotech.com> wrote:

> Packages are the best way, provided we can keep up and stay organized. Ken, I have no problems helping with packages but I would have to do a ton of research to get to a stage to be useful. I've been needing to learn how to do debian packaging and this could be my excuse to finally learn.
> 
> Brian Foster
> Endigo Computer LLC
> 
> Sent from a mobile device.
> 
> On Aug 11, 2012 3:13 PM, "Anthony Minessale" <anthony.minessale at gmail.com> wrote:
> Based on the inexplicable issues in cent6 working on debian packages might be very fruitful.  Talk to krice at freeswitch.org
> 
> On Aug 10, 2012 8:32 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
> I need to update the wiki, but the install script for Debian is at github.com/bdfoster/fs-debian-installer. It still gets the latest code on master. If you guys want to give me some feedback I can incorporate the proper changes. I stopped working on it because most are using centos and I am not going to maintain something no one uses. I have the time, I have the energy, but I need to know what people would like to see. Our internal script is based on this one but I cannot release those changes to the world.
> 
> Brian Foster
> Endigo Computer LLC
> 
> Sent from a mobile device.
> 
> On Aug 9, 2012 3:12 AM, "Miha" <miha at softnet.si> wrote:
> On 8/9/2012 8:36 AM, Peter Olsson wrote:
>> If you change to branch ?v1.2.stable?, the version will be 1.2.0.
>> 
>>  
>> 
>> The core devs have not yet sent out any official instructions, but I guess that if you want to continue to follow the main development (with sometimes unstable code), just continue to do as you do today. If you want to stick to the 1.2 version, and only get bugfixes etc. for that version (but no new features), change to the branch v1.2.stable.
>> 
>>  
>> 
>> I think there will be more official instructions soon, but this is that way I think it?s meant to work.
>> 
>>  
>> 
>> /Peter
>> 
>>  
>> 
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha
>> Skickat: den 9 augusti 2012 08:15
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>> 
>>  
>> 
>> Hi,
>> 
>> so my current version is 1.2 rc2. After make current I will have 1.2? I try it on test server but still FreeSWITCH Version 1.2.0-rc2+git~20120809T022906Z~3389f32363 (1.2.0-rc2; git at commit 3389f32363 on Thu, 09 Aug 2012 02:29:06 Z)
>> 
>> 
>> Thanks!
>> 
>> Miha
>> 
>> On 8/8/2012 5:02 PM, Ken Rice wrote:
>> 
>> the Primary repo is git.freeswitch.org/freeswitch.git this is the master repo, however, the github.com/freeswitch/freeswitch is an unofficial mirror...  updates to primary git repo should be pushed including tags and branches but we do not support those that wish to use other than official repos
>> 
>>  
>> 
>>  
>> 
>> K
>> 
>>  
>> 
>> On Wed, Aug 8, 2012 at 8:39 AM, Brian Foster <bdfoster at endigotech.com> wrote:
>> 
>> Ah, looks like they publish to github as well. Afaik the primary repo is git.freeswitch.org, and that is where I get everything.
>> 
>> Brian Foster
>> Endigo Computer LLC
>> 
>> Sent from a mobile device.
>> 
>> On Aug 8, 2012 8:41 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>> 
>> What is this then? https://github.com/freeswitch/freeswitch
>> 
>> 
>> Regards,
>> Ben Langfeld
>> 
>> 
>> On 8 August 2012 12:53, Brian Foster <bdfoster at endigotech.com> wrote:
>> 
>> FreeSWITCH isn't published on GitHub.
>> 
>> Brian Foster
>> Endigo Computer LLC
>> 
>> Sent from a mobile device.
>> 
>> On Aug 8, 2012 3:14 AM, "Stanislav Sinyagin" <ssinyagin at yahoo.com> wrote:
>> 
>> Ken,
>> 
>> care to push tags to Github as well?
>> 
>>  
>> 
>> From: Ken Rice <krice at freeswitch.org>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>> Sent: Tuesday, August 7, 2012 8:31 PM
>> Subject: Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>> 
>>  
>> 
>> yes its there... git pull --tags
>> 
>>  
>> 
>> On Tue, Aug 7, 2012 at 1:23 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>> 
>> I also checked tags, but didn't see it. Are you sure your branch is tracked upstream?
>> 
>>  
>> 
>> --Dave
>> 
>>  
>> 
>> From: Ken Rice [mailto:krice at freeswitch.org]
>> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
>> 
>> Sent: Tue, 07 Aug 2012 11:15:50 -0700
>> Subject: Re: [Freeswitch-users] Announcing FreeSWITCH 1.2
>> 
>> 
>> 
>> its Tagged, right now, we will be branching it shortly for tracking patches. Please note we had a couple of good users point out some smallish bugs so we're int he process of fixing those! so guys if you find anything wrong please open a jira, if you are at cluecon, find me and let me know the bug number
>> 
>>  
>> 
>> K
>> 
>> On Tue, Aug 7, 2012 at 1:12 PM, Dave R. Kompel <drk at drkngs.net> wrote:
>> 
>> Where is the branch so I can checkout local?
>> 
>>  
>> 
>> D:\fsnew>git branch -a
>> * master
>>   remotes/origin/FS-3432
>>   remotes/origin/FS-4062
>>   remotes/origin/HEAD -> origin/master
>>   remotes/origin/dingaling_video
>>   remotes/origin/master
>>   remotes/origin/smgfs
>>   remotes/origin/stable-test/freeswitch-1.2
>>   remotes/origin/swk/fs_test
>>   remotes/test/master
>> 
>> --Dave
>> 
>>  
>> 
>> From: Ken Rice [mailto:krice at freeswitch.org]
>> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org], freeswitch-dev at lists.freeswitch.org
>> Sent: Tue, 07 Aug 2012 09:03:34 -0700
>> 
>> 
>> Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
>> 
>> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!
>> 
>>  
>> 
>> Get your copy today at http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 ! 
>> 
>>  
>> 
>> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will continue to fix bugs and security issues. giving you a stable platform for at least one year.
>> 
>>  
>> 
>> Grab it today!
>> 
>>  
>> 
>> The FreeSWITCH Team 
>> 
>>  
>> 
>>  
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com/
>> 
>> 
>> /
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org/
>> http://wiki.freeswitch.org/
>> http://www.cluecon.com/
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>>  
>> 
>>  
>> 
>>  
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>>  
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>>  
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>>  
>> 
>> 
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>  
>> 
>> 
>>  
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>  
>> Join Us At ClueCon - Aug 7-9, 2012
>>  
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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:502354a432765721010175!
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> Join Us At ClueCon - Aug 7-9, 2012
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> @Peter,
> 
> thanks for all your explanation. 
> 
> Br,
> Miha
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lynn.nielson at greenseedtechnologies.com  Tue Aug 14 01:50:19 2012
From: lynn.nielson at greenseedtechnologies.com (Lynn Nielson)
Date: Mon, 13 Aug 2012 15:50:19 -0600
Subject: [Freeswitch-users] No Audio on external bridging
Message-ID: <5029769B.5080109@greenseedtechnologies.com>

We are building a dialer interface to our client app that calls our 
customers and then allows those customers to connect and autodial to 
their clients.  We are using these api commands to do this:

    originate [origination_caller_id_name
    ='[caller_id_name]',origination_caller_id_number=[caller_id_number],user_name=[a
    gent_id],park_after_bridge=true]sofia/gateway/vitelity-outbound/1[number_to_call
    ] &park

    uuid_audio [uuid1] start write level 0

    originate [origination_caller_id_nam
    e='[caller_id_name]',origination_caller_id_number=[caller_id_number],user_name=[
    agent_id]]sofia/gateway/vitelity-outbound/1[number_to_call] &park

    uuid_bridge [uuid1] [uuid2]


When the second call (leg-b) hangs up, we do not get a disconnect event 
(we believe this is due to the uuid_audio start).  We need that 
disconnect in order to launch the next call.

We are sending the uuid_audio [uuid1] start command because we do not 
have any media audio when we do the bridge.  We tried also sending a 
uuid_media off command but that has the consequence of dropping the 
first leg/call when the second call disconnects (does not honor the 
park_after_bridge).

This only happens when bridging two external calls.  If one of the legs 
is an internal extension, then we don't need the uuid_audio start or the 
uuid_media off and everything works fine.

Is there another way of doing this or a way for uuid_media off to honor 
the park_after_bridge condition?  Could we use fifo's instead of the 
park_after_bridge?

btw: we have also tried this using Flowroute as the gateway instead of 
Vitelity.


Thanks,

Lynn

-- 
--
Lynn Nielson
Green Seed Technologies / The RedX

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From lynn.nielson at greenseedtechnologies.com  Tue Aug 14 07:02:48 2012
From: lynn.nielson at greenseedtechnologies.com (Lynn Nielson)
Date: Mon, 13 Aug 2012 21:02:48 -0600
Subject: [Freeswitch-users] continuous ring back after rejecting call
Message-ID: <5029BFD8.3050700@greenseedtechnologies.com>

I have a problem when doing an api originate to an external number with 
a fifo park.  On some phones if the call is 'rejected' then freeswitch 
continuously rings back the number until it is answered. It has happened 
on cell phones, hard sip phones, and soft phones. Is there a way to 
always suppress the follow-up ring backs by setting a variable in the 
originate command (or something else I can set)?

Thanks,

Lynn



From eka at ekanet.net  Mon Aug 13 20:29:35 2012
From: eka at ekanet.net (Emrah)
Date: Mon, 13 Aug 2012 18:29:35 +0200
Subject: [Freeswitch-users] The domain_name variable and domains
	in	FreeSWITCH
In-Reply-To: <CAC-LwPPhp=d6w1XkAx_1pj1=J6JbjWRF9Yvx1Sno+GKC1KkPQQ@mail.gmail.com>
References: <20120813090751.GA15900@willow.kavun.ch>
	<CAC-LwPPhp=d6w1XkAx_1pj1=J6JbjWRF9Yvx1Sno+GKC1KkPQQ@mail.gmail.com>
Message-ID: <20120813162935.GA19250@willow.kavun.ch>

Thanks for your reply, I appreciate it.

I did read in the Wiki about how to configure a multi tenant instance, but it still does not tell me what function the domain name has.

I am looking for a technical explanation and some examples if possible.

Where and when is the domain name used in FreeSWITCH? Is it integrally part of a user's credentials? Will any aspect of the routing be affected by the actual domain name via 
DNS? (E.g.: how to differentiate on a core level users calling joe at a.net and joe at b.net where a.net and b.net point to the same SIP IP? I am not talking about a conditional 
routing in the dialplan here.)

Thanks and best regards,
Emrah

From: Vik Killa <vipkilla at gmail.com>
Sent: Mon, Aug 13, 2012 at 11:52:10AM -0400
Subject: Re: [Freeswitch-users] The domain_name variable and domains in FreeSWITCH

> If properly configured you can have each domain operate as a different
> tenant on the server, so they will be separate PBXs.
> 
> On Mon, Aug 13, 2012 at 5:07 AM, Emrah <eka at ekanet.net> wrote:
> > Hello all,
> >
> > I have been an Asterisk user for almost 10 years... God I'm glad I took the leap and tried FS!
> > Congratulations to all of you for the amazing dev you've put together.
> >
> > I have been trying to understand in depth how the domain_name variable is used throughout FS. Or more in general, what defines a domain name and how is it used?
> >
> > It doesn't seem that FS is domain sensitive for registrations / authenticated calls. If both sip.example.com and fs.example.com are IN A 1.2.3.4, I can register on both domains with no
> > distinction whatsoever.
> >
> > How does that work?
> >
> > Best to you and thanks for your precious answers.
> > --
> > Emrah
> > <eka at ekanet.net>
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Emrah 
<eka at ekanet.net>


From govoiper at gmail.com  Tue Aug 14 07:56:15 2012
From: govoiper at gmail.com (SamyGo)
Date: Tue, 14 Aug 2012 08:56:15 +0500
Subject: [Freeswitch-users] can I set the "api originate" command to use
	dialplan?
In-Reply-To: <CAKV-jFx=TVu3rxk8N49P_DQ1CSTM9cmXNuoxV-bMsfO1anw2eQ@mail.gmail.com>
References: <50293975.9090401@daitangroup.com>
	<CAKV-jFx=TVu3rxk8N49P_DQ1CSTM9cmXNuoxV-bMsfO1anw2eQ@mail.gmail.com>
Message-ID: <CAJUJwtgqLmU1gu_wUNCM1JNroHNXnf8jJwbobJsWJJNqe2dmcg@mail.gmail.com>

Though Brain already spoke on this, just to add. If you are thinking of
handing your call over to the dialplan and do some processing before
sending out call then you need to use loopback channel instead of sofia.
That way you can dial the pstn number within your dialplan and do something
there.
Dialplan needs to be in place obviously.

Otherwise I dont think there is any mechanism or key-word you can put in
that field and it do the LCR or distribution or failover the carrier it
self.

BR
Sammy G.
On Aug 14, 2012 4:11 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> that's up to your script to provide. You'really already at the lowest
> level to make a call via event socket.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Aug 13, 2012 1:33 PM, "Leonardo" <leonardo at daitangroup.com> wrote:
>
>>  Hi.
>> I'm using the command "originate" to make a call via socket interface:
>>
>> api originate sofia/external/MY_PSTN_NUMBER at PSTN_GATEWAY &park()
>> api originate sofia/external/MY_PSTN_NUMBER at PSTN_GATEWAY&bridge(originate sofia/external/OTHER_PSTN_NUMBER@
>> PSTN_GATEWAY)
>>
>> Is there a way to omit the "PSTN_GATEWAY" and force the Freeswitch
>> figure out which one to use based on PSTN_NUMBER/dialplan?
>>
>> Thanks i advance,
>> Leo
>>
>>  --
>>
>> Leonardo N. S. Pereira, *Software Engineer*
>> T:+55.19.3112-1200 ext. 1283*|*F:+55.19.3207-1437
>> *Daitan**Group**|**www.daitangroup.com* <http://www.daitangroup.com/>*|**Highly
>> Reliable Outsourcing. Value Added Services Worldwide.*
>> *Privileged and confidential. If this message has been received in
>> error, please notify sender and delete it immediately.*
>> *Conte?do confidencial. Se esta mensagem foi recebida por engano, favor
>> avisar o remetente e apag?-la imediatamente.*
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bdfoster at endigotech.com  Tue Aug 14 08:53:35 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 14 Aug 2012 00:53:35 -0400
Subject: [Freeswitch-users] The domain_name variable and domains in
	FreeSWITCH
In-Reply-To: <20120813162935.GA19250@willow.kavun.ch>
References: <20120813090751.GA15900@willow.kavun.ch>
	<CAC-LwPPhp=d6w1XkAx_1pj1=J6JbjWRF9Yvx1Sno+GKC1KkPQQ@mail.gmail.com>
	<20120813162935.GA19250@willow.kavun.ch>
Message-ID: <CAKV-jFy-UAcjo6MaVyba1Um-yPuDV=uUaRFXLzEuo02MX26fxA@mail.gmail.com>

It's all in the XML. Believe me, it works and works quite well :)

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 13, 2012 11:35 PM, "Emrah" <eka at ekanet.net> wrote:

> Thanks for your reply, I appreciate it.
>
> I did read in the Wiki about how to configure a multi tenant instance, but
> it still does not tell me what function the domain name has.
>
> I am looking for a technical explanation and some examples if possible.
>
> Where and when is the domain name used in FreeSWITCH? Is it integrally
> part of a user's credentials? Will any aspect of the routing be affected by
> the actual domain name via
> DNS? (E.g.: how to differentiate on a core level users calling joe at a.netand
> joe at b.net where a.net and b.net point to the same SIP IP? I am not
> talking about a conditional
> routing in the dialplan here.)
>
> Thanks and best regards,
> Emrah
>
> From: Vik Killa <vipkilla at gmail.com>
> Sent: Mon, Aug 13, 2012 at 11:52:10AM -0400
> Subject: Re: [Freeswitch-users] The domain_name variable and domains in
> FreeSWITCH
>
> > If properly configured you can have each domain operate as a different
> > tenant on the server, so they will be separate PBXs.
> >
> > On Mon, Aug 13, 2012 at 5:07 AM, Emrah <eka at ekanet.net> wrote:
> > > Hello all,
> > >
> > > I have been an Asterisk user for almost 10 years... God I'm glad I
> took the leap and tried FS!
> > > Congratulations to all of you for the amazing dev you've put together.
> > >
> > > I have been trying to understand in depth how the domain_name variable
> is used throughout FS. Or more in general, what defines a domain name and
> how is it used?
> > >
> > > It doesn't seem that FS is domain sensitive for registrations /
> authenticated calls. If both sip.example.com and fs.example.com are IN A
> 1.2.3.4, I can register on both domains with no
> > > distinction whatsoever.
> > >
> > > How does that work?
> > >
> > > Best to you and thanks for your precious answers.
> > > --
> > > Emrah
> > > <eka at ekanet.net>
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > Join Us At ClueCon - Aug 7-9, 2012
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Emrah
> <eka at ekanet.net>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From covici at ccs.covici.com  Tue Aug 14 10:16:48 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 14 Aug 2012 02:16:48 -0400
Subject: [Freeswitch-users] fs crashs after more then 100 dtmf events on
	a call
In-Reply-To: <Pine.LNX.4.64.1208140044060.21902@1a-8140.antagus.de>
References: <1FFF97C269757C458224B7C895F35F1514327E@cantor.std.visionutv.se>,
	<Pine.LNX.4.64.1208131621500.19780@1a-8140.antagus.de>
	<E19ECE61-5F21-48B6-8722-D510B05A74C0@visionutveckling.se>
	<Pine.LNX.4.64.1208140044060.21902@1a-8140.antagus.de>
Message-ID: <28262.1344925008@ccs.covici.com>

I use pastebin all the time with lynx with no problems -- write me off
list for more information.

Thomas Hoellriegel <admin at blindi.net> wrote:

> Hi Peter,
> sorry, im.m a blind user. I can only use a textbased browser lynx on
> linux.
> The pastebin interface can.t be  used with text only browsers.
> 
> I have store the cord dump on my website:
> You can download at:
> www.blindi.net/core.gz
> wget wget www.blindi.net/core.gz
> 
> 
> fs breaks with a segfault.
> 
> Thanks.
> 
> 
> ---------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 vaad.fabi at gmail.com  Tue Aug 14 10:50:57 2012
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Tue, 14 Aug 2012 09:50:57 +0300
Subject: [Freeswitch-users] jira improvements
In-Reply-To: <BAY156-W466967C2B4CE10A8B3AE2CD4B10@phx.gbl>
References: <BAY156-W11F916CCDCFC6B2A6B3E2DD4CC0@phx.gbl>,
	<CALn233rwaSGbtbcOV8sJPRA561T9MoWooBRYk4E=RNuvq9UM+Q@mail.gmail.com>
	<BAY156-W466967C2B4CE10A8B3AE2CD4B10@phx.gbl>
Message-ID: <5029F551.4070606@gmail.com>

  Please add to jira into field "Component\s" value mod_rtmp.
Thx


-- 
Best Regards,
Vadim F.



From krice at freeswitch.org  Tue Aug 14 11:13:21 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 14 Aug 2012 02:13:21 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.2.1 has been rolled...
Message-ID: <CC4F64C1.3C00B%krice@freeswitch.org>

FreeSWITCH 1.2.1 is Here!!!

Tarball is on files.freeswitch.org as usual.

Also Debian users, check out the debian repo on
http://files.freeswitch.org/repo  More info on this coming soon, currently
this continues nightly builds from the dev branch but will soon also contain
a stable repo... 

As usual, any issues please report them via jira.freeswitch.org and please
tag them with the correct version!


K 
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From lists at kavun.ch  Tue Aug 14 07:58:15 2012
From: lists at kavun.ch (Emrah)
Date: Tue, 14 Aug 2012 05:58:15 +0200
Subject: [Freeswitch-users] The domain_name variable and domains in
	FreeSWITCH
In-Reply-To: <CAKV-jFzsCXq9S69cetFbgAvb=yWswUNXyk+O1YdYs52aXydt3Q@mail.gmail.com>
References: <20120813090751.GA15900@willow.kavun.ch>
	<CAKV-jFzsCXq9S69cetFbgAvb=yWswUNXyk+O1YdYs52aXydt3Q@mail.gmail.com>
Message-ID: <0E00B7CD-FDD6-4357-88C3-B8A68A4FEE14@kavun.ch>

I know it sounds like a trivial question. I just wanted to make sure I had a clear understanding of the implication of a domain name throughout FS.

Asterisk in SIP will reject calls made to an unknown domain. E.g.: sip.example-a.com and sip.example-b.com both commonly IN A 1.2.3.4.
If sip.conf only contains:
domain=1.2.3.4
domain=sip.example-a.com
Calls made to sip.example-b.com will be rejected.

You can also use the same setting to force a call into a particular context, so that 11 at conference.sip.example-a.com will land in a context defined for conferences.

I thought domain names in FS had some more substantial implications and wanted to get a good grasp of them.

All the best and thanks again for your input,
Emrah

On Aug 14, 2012, at 3:37 AM, Brian Foster <bdfoster at endigotech.com> wrote:

> This falls into the multi tenant catagory. Please do a search on the wiki for multitenant there are guides on how to change that behavior. Its a relatively simple process. However you need to change certain aspects like file locations to accomidate multiple tenants. By default $$domain is generated as the local ipv4 address.
> 
> Brian Foster
> Endigo Computer LLC
> 
> Sent from a mobile device.
> 
> On Aug 13, 2012 9:43 AM, "Emrah" <eka at ekanet.net> wrote:
> Hello all,
> 
> I have been an Asterisk user for almost 10 years... God I'm glad I took the leap and tried FS!
> Congratulations to all of you for the amazing dev you've put together.
> 
> I have been trying to understand in depth how the domain_name variable is used throughout FS. Or more in general, what defines a domain name and how is it used?
> 
> It doesn't seem that FS is domain sensitive for registrations / authenticated calls. If both sip.example.com and fs.example.com are IN A 1.2.3.4, I can register on both domains with no
> distinction whatsoever.
> 
> How does that work?
> 
> Best to you and thanks for your precious answers.
> --
> Emrah
> <eka at ekanet.net>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From curriegrad2004 at gmail.com  Tue Aug 14 11:20:23 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 14 Aug 2012 00:20:23 -0700
Subject: [Freeswitch-users] FreeSWITCH 1.2.1 has been rolled...
In-Reply-To: <CC4F64C1.3C00B%krice@freeswitch.org>
References: <CC4F64C1.3C00B%krice@freeswitch.org>
Message-ID: <CAMWiMy-md2CT3dPbBPa2cQcSG09hiCyo19PBdiN7H6r_J5e1kA@mail.gmail.com>

1.2.1 Should be known as 1.2.0 SP1 :P

On Tue, Aug 14, 2012 at 12:13 AM, Ken Rice <krice at freeswitch.org> wrote:
> FreeSWITCH 1.2.1 is Here!!!
>
> Tarball is on files.freeswitch.org as usual.
>
> Also Debian users, check out the debian repo on
> http://files.freeswitch.org/repo  More info on this coming soon, currently
> this continues nightly builds from the dev branch but will soon also contain
> a stable repo...
>
> As usual, any issues please report them via jira.freeswitch.org and please
> tag them with the correct version!
>
>
> K
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From gerald.weber at besharp.at  Tue Aug 14 11:23:18 2012
From: gerald.weber at besharp.at (Gerald Weber)
Date: Tue, 14 Aug 2012 07:23:18 +0000
Subject: [Freeswitch-users] jira improvements
In-Reply-To: <5029F551.4070606@gmail.com>
References: <BAY156-W11F916CCDCFC6B2A6B3E2DD4CC0@phx.gbl>,
	<CALn233rwaSGbtbcOV8sJPRA561T9MoWooBRYk4E=RNuvq9UM+Q@mail.gmail.com>
	<BAY156-W466967C2B4CE10A8B3AE2CD4B10@phx.gbl>
	<5029F551.4070606@gmail.com>
Message-ID: <D1925DECD316894DAADB84076C1B7D827E286645@exchange.besharp.at>

And mod_snom please :)

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von vaad.fabi at gmail.com
Gesendet: Dienstag, 14. August 2012 08:51
An: anthony.minessale at gmail.com
Cc: freeswitch-users at lists.freeswitch.org
Betreff: [Freeswitch-users] jira improvements

  Please add to jira into field "Component\s" value mod_rtmp.
Thx


-- 
Best Regards,
Vadim F.


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

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


From evgeniy at bestnet.kharkov.ua  Tue Aug 14 12:30:27 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Tue, 14 Aug 2012 11:30:27 +0300
Subject: [Freeswitch-users] Error while make current
Message-ID: <502A0CA3.80205@bestnet.kharkov.ua>

Hi to all. I tried to update my FS, but got this message:

make[1]: Entering directory `/usr/local/src/freeswitch'
Pulling updates...
remote: Counting objects: 335, done.
remote: Compressing objects: 100% (95/95), done.
remote: Total 244 (delta 192), reused 187 (delta 148)
Receiving objects: 100% (244/244), 109.48 KiB | 175 KiB/s, done.
Resolving deltas: 100% (192/192), completed with 62 local objects.
 From git://git.freeswitch.org/freeswitch
    aad07c6..3d9d42b  master     -> origin/master
    9732411..f2f4f4a  v1.2.stable -> origin/v1.2.stable
Updating 9ac586a..3d9d42b
error: Your local changes to 'libs/spandsp/INSTALL' would be overwritten 
by merge.  Aborting.
Please, commit your changes or stash them before you can merge.
make[1]: *** [update] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch'
make: *** [current] Error 2


-- 
Evgeniy Movlyan,
BestNet Ltd.


From peter.olsson at visionutveckling.se  Tue Aug 14 12:51:30 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 14 Aug 2012 08:51:30 +0000
Subject: [Freeswitch-users] Error while make current
Message-ID: <1FFF97C269757C458224B7C895F35F151444A7@cantor.std.visionutv.se>

"git checkout -- libs/spandsp/INSTALL" - before make current

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Evgeniy Movlyan
Skickat: den 14 augusti 2012 10:30
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Error while make current

Hi to all. I tried to update my FS, but got this message:

make[1]: Entering directory `/usr/local/src/freeswitch'
Pulling updates...
remote: Counting objects: 335, done.
remote: Compressing objects: 100% (95/95), done.
remote: Total 244 (delta 192), reused 187 (delta 148) Receiving objects: 100% (244/244), 109.48 KiB | 175 KiB/s, done.
Resolving deltas: 100% (192/192), completed with 62 local objects.
 From git://git.freeswitch.org/freeswitch
    aad07c6..3d9d42b  master     -> origin/master
    9732411..f2f4f4a  v1.2.stable -> origin/v1.2.stable Updating 9ac586a..3d9d42b
error: Your local changes to 'libs/spandsp/INSTALL' would be overwritten by merge.  Aborting.
Please, commit your changes or stash them before you can merge.
make[1]: *** [update] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch'
make: *** [current] Error 2


-- 
Evgeniy Movlyan,
BestNet Ltd.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
FreeSWITCH-users at 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:502a0a7d32761616232089!



From admin at blindi.net  Tue Aug 14 14:15:27 2012
From: admin at blindi.net (Thomas Hoellriegel)
Date: Tue, 14 Aug 2012 12:15:27 +0200 (CEST)
Subject: [Freeswitch-users] fs crashs after more then 100 dtmf events on
 a call
In-Reply-To: <28262.1344925008@ccs.covici.com>
References: <1FFF97C269757C458224B7C895F35F1514327E@cantor.std.visionutv.se>, 
	<Pine.LNX.4.64.1208131621500.19780@1a-8140.antagus.de>
	<E19ECE61-5F21-48B6-8722-D510B05A74C0@visionutveckling.se>
	<Pine.LNX.4.64.1208140044060.21902@1a-8140.antagus.de>
	<28262.1344925008@ccs.covici.com>
Message-ID: <Pine.LNX.4.64.1208141205390.25828@1a-8140.antagus.de>

Am 14.08.12 um 02:16 schrieb covici at ccs.covici.com:

> I use pastebin all the time with lynx with no problems -- write me off
> list for more information.

How can you pate or upload a logfile?
I using debian squeeze in a text only console with a brailledisplay.

I can cut an paste with the brailledisplay only 25x80 blocks.
This is my problem.
I have tested with screen. Screen don.t pate a big logfile correctly.

I have create the copedump from fs. This is not the problem.
The problem ist: to paste the needed informations.

Thanks.

------------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste f?r blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc

From covici at ccs.covici.com  Tue Aug 14 17:24:18 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 14 Aug 2012 09:24:18 -0400
Subject: [Freeswitch-users] fs crashs after more then 100 dtmf events on
	a call
In-Reply-To: <Pine.LNX.4.64.1208141205390.25828@1a-8140.antagus.de>
References: <1FFF97C269757C458224B7C895F35F1514327E@cantor.std.visionutv.se>,
	<Pine.LNX.4.64.1208131621500.19780@1a-8140.antagus.de>
	<E19ECE61-5F21-48B6-8722-D510B05A74C0@visionutveckling.se>
	<Pine.LNX.4.64.1208140044060.21902@1a-8140.antagus.de>
	<28262.1344925008@ccs.covici.com>
	<Pine.LNX.4.64.1208141205390.25828@1a-8140.antagus.de>
Message-ID: <22983.1344950658@ccs.covici.com>

If you are using lynx, and you are on an edit field,  you can do
control-x e and this will get you into your preferred editor -- please
make sure you have selected that in the lynx options and then you can do
whatever you want.

Hope this helps.

Thomas Hoellriegel <admin at blindi.net> wrote:

> Am 14.08.12 um 02:16 schrieb covici at ccs.covici.com:
> 
> > I use pastebin all the time with lynx with no problems -- write me off
> > list for more information.
> 
> How can you pate or upload a logfile?
> I using debian squeeze in a text only console with a brailledisplay.
> 
> I can cut an paste with the brailledisplay only 25x80 blocks.
> This is my problem.
> I have tested with screen. Screen don.t pate a big logfile correctly.
> 
> I have create the copedump from fs. This is not the problem.
> The problem ist: to paste the needed informations.
> 
> Thanks.
> 
> ------------------
> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
> http://www.blindi.net/callback
> homepage: http://www.blindi.net
> blinde-misc mailingliste f?r blinde. anmeldung unter:
> http://www.blindi.net/mailman/listinfo/blinde-misc
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 itispip-qq at hotmail.com  Tue Aug 14 14:34:02 2012
From: itispip-qq at hotmail.com (Pip Live)
Date: Tue, 14 Aug 2012 18:34:02 +0800
Subject: [Freeswitch-users] Does chatplan support javascript?
In-Reply-To: <CAKbxfG9aUwUth9MVj49MD3WWWj6=rfB9-DeZUq-26LsjgbdQ-w@mail.gmail.com>
References: <BLU169-W2039DAEFAF1A65AF380A7A8CB30@phx.gbl>
	<CAKbxfG9aUwUth9MVj49MD3WWWj6=rfB9-DeZUq-26LsjgbdQ-w@mail.gmail.com>
Message-ID: <BLU0-SMTP1401B651B12AD4A5B9087988CB70@phx.gbl>

Hi, new Javascript interface in chatplan works well; But strangely I cannot
call js function "email", every attempt ended up "Reference error, email is
not defined". Call to other js fucntions no problem.

Is this function missed in src?

Pip

2012/8/11 Anthony Minessale <anthony.minessale at gmail.com>

> its entirely possible but does not yet exist.
> I pushed a patch that may enable it, but i did not have a way to test
> it atm.  So you will have to try it for me and see what happens.
>
> try git HEAD
>
>
>
> On Fri, Aug 10, 2012 at 5:35 AM, ?? <itispip-qq at hotmail.com> wrote:
> > Hi FS Gurus,
> >
> > I'm new to freeswitch. Trying to use javascript in chatplan, but get
> error
> > code c:716 Invalid chat application interface [javascript]!
> >
> > Same fucntion call in Lua is OK. So looking chatplan not support
> javascript?
> >
> > If yes, any plan to add javascript to chatplan? Chatplan mainly dealing
> with
> > text manipulation, Lua is a bit weak at string handling.
> >
> > /rgds, Pip
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From wingcomm at hotmail.com  Tue Aug 14 15:59:19 2012
From: wingcomm at hotmail.com (R W)
Date: Tue, 14 Aug 2012 07:59:19 -0400
Subject: [Freeswitch-users] Calls from SRTP Clients to non-SRTP clients
Message-ID: <BAY156-W3667570B62D531ABCA6E7FD4B70@phx.gbl>


Freeswitch-users,
I'm trying to migrate towards SRTP on our handsets. However, some of my softclients (Bria for the iPhone and Windows/Mac OS X) only support "all-or-nothing" SRTP. Meaning, the caller must support SRTP or must not support it. I know Bria on the iPhone supports an "if supported" mode, however, Counterpath's Bria for the iPhone implementation of this generates errors on outbound calls to FreeSWITCH.
So, my question is, is there a way to force FreeSWITCH to establish an SRTP call to clients when the originating client does not support SRTP?
-Rob 		 	   		  
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From mitch.capper at gmail.com  Tue Aug 14 18:35:25 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 14 Aug 2012 07:35:25 -0700
Subject: [Freeswitch-users] Calls from SRTP Clients to non-SRTP clients
In-Reply-To: <BAY156-W3667570B62D531ABCA6E7FD4B70@phx.gbl>
References: <BAY156-W3667570B62D531ABCA6E7FD4B70@phx.gbl>
Message-ID: <CALn233ozDnC5mpzuwC5K8QEjF8P4Y5hNDO8F4qD92w3o_K-mjA@mail.gmail.com>

Of course, SRTP is not an endpoint to endpoint protocol but generally
is between the client and the server anyway so any legs you want can
use SRTP  for outgoing just obviously register using TLS and for
incoming just set the secure media flags to the users who are supposed
to be encrypted.

~mitch

On Tue, Aug 14, 2012 at 4:59 AM, R W <wingcomm at hotmail.com> wrote:
> Freeswitch-users,
>
> I'm trying to migrate towards SRTP on our handsets. However, some of my
> softclients (Bria for the iPhone and Windows/Mac OS X) only support
> "all-or-nothing" SRTP. Meaning, the caller must support SRTP or must not
> support it. I know Bria on the iPhone supports an "if supported" mode,
> however, Counterpath's Bria for the iPhone implementation of this generates
> errors on outbound calls to FreeSWITCH.
>
> So, my question is, is there a way to force FreeSWITCH to establish an SRTP
> call to clients when the originating client does not support SRTP?
>
> -Rob
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Tue Aug 14 20:10:28 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 14 Aug 2012 11:10:28 -0500
Subject: [Freeswitch-users] Does chatplan support javascript?
In-Reply-To: <BLU0-SMTP1401B651B12AD4A5B9087988CB70@phx.gbl>
References: <BLU169-W2039DAEFAF1A65AF380A7A8CB30@phx.gbl>
	<CAKbxfG9aUwUth9MVj49MD3WWWj6=rfB9-DeZUq-26LsjgbdQ-w@mail.gmail.com>
	<BLU0-SMTP1401B651B12AD4A5B9087988CB70@phx.gbl>
Message-ID: <CAKbxfG9GqzaaRj+bHV6o73QCas-y4bxSgQcA9m9TZv1SKYVMTg@mail.gmail.com>

that did not exist,

I just added it but if you find another obstacle after that you're on your own.


On Tue, Aug 14, 2012 at 5:34 AM, Pip Live <itispip-qq at hotmail.com> wrote:
> Hi, new Javascript interface in chatplan works well; But strangely I cannot
> call js function "email", every attempt ended up "Reference error, email is
> not defined". Call to other js fucntions no problem.
>
> Is this function missed in src?
>
> Pip
>
> 2012/8/11 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> its entirely possible but does not yet exist.
>> I pushed a patch that may enable it, but i did not have a way to test
>> it atm.  So you will have to try it for me and see what happens.
>>
>> try git HEAD
>>
>>
>>
>> On Fri, Aug 10, 2012 at 5:35 AM, ?? <itispip-qq at hotmail.com> wrote:
>> > Hi FS Gurus,
>> >
>> > I'm new to freeswitch. Trying to use javascript in chatplan, but get
>> > error
>> > code c:716 Invalid chat application interface [javascript]!
>> >
>> > Same fucntion call in Lua is OK. So looking chatplan not support
>> > javascript?
>> >
>> > If yes, any plan to add javascript to chatplan? Chatplan mainly dealing
>> > with
>> > text manipulation, Lua is a bit weak at string handling.
>> >
>> > /rgds, Pip
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > Join Us At ClueCon - Aug 7-9, 2012
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Aug 14 20:33:06 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 14 Aug 2012 09:33:06 -0700
Subject: [Freeswitch-users] Collateral Damage
In-Reply-To: <CAAxwcqsPcL7yKXfbo5VY7ZYMXn2xF3h5vya=2T7DhiT+OXU+og@mail.gmail.com>
References: <CAAxwcqvqEUoNnbG+kiPJeo0Ohx20HVtE0oz6GyjLyJrF7uB4cQ@mail.gmail.com>
	<960588CE0EB9415696CF8173741807C6@gmail.com>
	<CAAxwcqsPcL7yKXfbo5VY7ZYMXn2xF3h5vya=2T7DhiT+OXU+og@mail.gmail.com>
Message-ID: <CAKzWOxUFi1PgVJ2a30FV2C3WvScEE-81ATFon11N3ycUbsg+_A@mail.gmail.com>

On Fri, Aug 10, 2012 at 8:56 PM, Vishal Kakkar <vishal.kakkar at gmail.com>wrote:

> Thanks Chad.. Gr8 Help!!


For posterity's sake, you can also get a copy of the currently running XML
configs from fs_cli with this command:

xml_locate root

It will dump everything (with a few extra blank lines) and you can go from
there...
-MC
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From rnbrady at gmail.com  Tue Aug 14 21:41:03 2012
From: rnbrady at gmail.com (Richard Brady)
Date: Tue, 14 Aug 2012 18:41:03 +0100
Subject: [Freeswitch-users] Calls from SRTP Clients to non-SRTP clients
In-Reply-To: <BAY156-W3667570B62D531ABCA6E7FD4B70@phx.gbl>
References: <BAY156-W3667570B62D531ABCA6E7FD4B70@phx.gbl>
Message-ID: <CAN5esQJUoQ2RFgAih=Wd59V=dMuh3od1bobnw4jAA_ymM1btoQ@mail.gmail.com>

> is there a way to force FreeSWITCH to establish an SRTP call to clients
when the originating client does not support SRTP?

This should work by default, assuming you are setting sip_secure_media in
the appropriate place.

FreeSWITCH should negotiate both channels (legs) independently. So if the
A-end has no SRTP, that should not prevent FreeSWITCH from sending a INVITE
to the B-end with SRTP specified (i.e. SAVP in the SDP with a crypto
attribute).

I think "all or nothing" doesn't imply both ends of the call, it implies
all calls or none of the calls calls. So an inbound or outbound call
without SRTP will be rejected. Hope this makes sense.

However, in the default dialplan there is a condition that will cause
FreeSWITCH to implement such a policy. It is commented out by default:

      <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>

So if you uncommented that export line you would experience the behaviour
you described.

Assuming you have not done that, could it be that Bria is simply rejecting
any INVITE with SDP that does not contain an SAVP entry with a crypto
attribute? If this was the case you would find all inbound call to that
extension failing.

Actually I wonder if this is what happened and then caused you to uncomment
the line above, which has led you to your conclusion, as this would cause
only calls coming from SRTP devices to work. If so, you'd want to comment
it out again and find a different way to create a group for all users with
SRTP devices and use a dialplan condition to decide whether or not to
export sip_secure_media=true.

Alternatively you could try for some sort of fall-back mechanism but you'd
have to think carefully about this to make it secure and/or stable.

Good luck!

Richard

PS: In your first paragraph, did you mean Bria for iPhone in both cases?
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From phil.kim at valiant.com  Wed Aug 15 00:08:59 2012
From: phil.kim at valiant.com (Phil Kim)
Date: Tue, 14 Aug 2012 16:08:59 -0400
Subject: [Freeswitch-users] Any good Canadian SIP carriers
Message-ID: <C69FF59885AC464896A4B68500B6E974098CF5E2@VAL-SV-EXCH-01.valiant.com>

Hello all,

 

I am looking for a good SIP carrier (provider) in Canada. We already
have a carrier in U.S. but they do not offer good rates for calls coming
over from Canada. 

 

Any inputs will be appreciated. 

 

Thank you.

 

Best regards,

 

Phil H. Kim

 

The information contained in this e-mail message, and any attachment
thereto, is the property of Valiant and is confidential and may not be
disclosed without our express permission. If you are not the intended
recipient or an employee or agent responsible for delivering this message to
the intended recipient, you are hereby notified that you have received this
message in error and that any review, dissemination, distribution or copying
of this message, or any attachment thereto, in whole or in part, is strictly
prohibited. If you have received this message in error, please immediately
notify us by telephone, fax or e-mail and delete the message and all of its
attachments. 

Thank you. 

Every effort is made to keep our network free from viruses. You should,
however, review this e-mail message, as well as any attachment thereto,
for viruses. We take no responsibility and have no liability for any
computer virus which may be transferred via this e-mail message.
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From jkomar at jbox.ca  Wed Aug 15 00:21:25 2012
From: jkomar at jbox.ca (Komar, Jason)
Date: Tue, 14 Aug 2012 14:21:25 -0600
Subject: [Freeswitch-users] Any good Canadian SIP carriers
In-Reply-To: <C69FF59885AC464896A4B68500B6E974098CF5E2@VAL-SV-EXCH-01.valiant.com>
References: <C69FF59885AC464896A4B68500B6E974098CF5E2@VAL-SV-EXCH-01.valiant.com>
Message-ID: <CA+zebiY7XtP_p0r1K8V6+ckrxXkiLjUAHQzHMaQv+_wLFbN0=Q@mail.gmail.com>

We use voip.ms. We've had good results. They don't support T38 faxing
though, so if you want fax you have to use an ATA.

Jason

On Tue, Aug 14, 2012 at 2:08 PM, Phil Kim <phil.kim at valiant.com> wrote:

> ** **
>
> Hello all,****
>
> ** **
>
> I am looking for a good SIP carrier (provider) in ****Canada****. We
> already have a carrier in **U.S.** but they do not offer good rates for
> calls coming over from ****Canada****. ****
>
> ** **
>
> Any inputs will be appreciated. ****
>
> ** **
>
> Thank you.****
>
> ** **
>
> Best regards,****
>
> ** **
>
> Phil H. Kim****
>
> ** **
>
> The information contained in this e-mail message, and any attachment
> thereto, is the property of Valiant and is confidential and may not be
> disclosed without our express permission. If you are not the intended
> recipient or an employee or agent responsible for delivering this message
> to
> the intended recipient, you are hereby notified that you have received this
> message in error and that any review, dissemination, distribution or
> copying
> of this message, or any attachment thereto, in whole or in part, is
> strictly
> prohibited. If you have received this message in error, please immediately
> notify us by telephone, fax or e-mail and delete the message and all of its
> attachments.
>
> Thank you.
>
> Every effort is made to keep our network free from viruses. You should,
> however, review this e-mail message, as well as any attachment thereto,
> for viruses. We take no responsibility and have no liability for any
> computer virus which may be transferred via this e-mail message.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From claytondus at gmail.com  Wed Aug 15 00:26:37 2012
From: claytondus at gmail.com (Clayton Davis)
Date: Tue, 14 Aug 2012 15:26:37 -0500
Subject: [Freeswitch-users] Any good Canadian SIP carriers
In-Reply-To: <C69FF59885AC464896A4B68500B6E974098CF5E2@VAL-SV-EXCH-01.valiant.com>
References: <C69FF59885AC464896A4B68500B6E974098CF5E2@VAL-SV-EXCH-01.valiant.com>
Message-ID: <CADaCfqWr=cExB0WN8OsTbV2M1Hq-ea2Vxb750U+kP7qfAxN-hg@mail.gmail.com>

I've used LES.NET since 2004 and have been very pleased with them.  Every
account ($1.99 CDN/mo) gets a free 24 channel DID.  $0.015 CDN termination
to US/CA.

-cd

2012/8/14 Phil Kim <phil.kim at valiant.com>

> ** **
>
> Hello all,****
>
> ** **
>
> I am looking for a good SIP carrier (provider) in ****Canada****. We
> already have a carrier in **U.S.** but they do not offer good rates for
> calls coming over from ****Canada****. ****
>
> ** **
>
> Any inputs will be appreciated. ****
>
> ** **
>
> Thank you.****
>
> ** **
>
> Best regards,****
>
> ** **
>
> Phil H. Kim****
>
> ** **
>
> The information contained in this e-mail message, and any attachment
> thereto, is the property of Valiant and is confidential and may not be
> disclosed without our express permission. If you are not the intended
> recipient or an employee or agent responsible for delivering this message
> to
> the intended recipient, you are hereby notified that you have received this
> message in error and that any review, dissemination, distribution or
> copying
> of this message, or any attachment thereto, in whole or in part, is
> strictly
> prohibited. If you have received this message in error, please immediately
> notify us by telephone, fax or e-mail and delete the message and all of its
> attachments.
>
> Thank you.
>
> Every effort is made to keep our network free from viruses. You should,
> however, review this e-mail message, as well as any attachment thereto,
> for viruses. We take no responsibility and have no liability for any
> computer virus which may be transferred via this e-mail message.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
__________________
Clayton Davis
ADTRAN
claytondus at gmail.com
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From sos at sokhapkin.dyndns.org  Wed Aug 15 00:55:50 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 14 Aug 2012 16:55:50 -0400
Subject: [Freeswitch-users] Any good Canadian SIP carriers
In-Reply-To: <CADaCfqWr=cExB0WN8OsTbV2M1Hq-ea2Vxb750U+kP7qfAxN-hg@mail.gmail.com>
References: <C69FF59885AC464896A4B68500B6E974098CF5E2@VAL-SV-EXCH-01.valiant.com>
	<CADaCfqWr=cExB0WN8OsTbV2M1Hq-ea2Vxb750U+kP7qfAxN-hg@mail.gmail.com>
Message-ID: <5905014.q0LB2PQHnr@sos>

 1.5c/m to USA/Canada is pretty high now-days. My rates at callwithus.com are 
0.95c/m USA and 0.5c/m Canada.

On Tuesday 14 August 2012 15:26:37 Clayton Davis wrote:
> I've used LES.NET since 2004 and have been very pleased with them.  Every
> account ($1.99 CDN/mo) gets a free 24 channel DID.  $0.015 CDN termination
> to US/CA.
> 
> -cd
> 
> 2012/8/14 Phil Kim <phil.kim at valiant.com>
> 
> > ** **
> > 
> > Hello all,****
> > 
> > ** **
> > 
> > I am looking for a good SIP carrier (provider) in ****Canada****. We
> > already have a carrier in **U.S.** but they do not offer good rates for
> > calls coming over from ****Canada****. ****
> > 
> > ** **
> > 
> > Any inputs will be appreciated. ****
> > 
> > ** **
> > 
> > Thank you.****
> > 
> > ** **
> > 
> > Best regards,****
> > 
> > ** **
> > 
> > Phil H. Kim****
> > 
> > ** **
> > 
> > The information contained in this e-mail message, and any attachment
> > thereto, is the property of Valiant and is confidential and may not be
> > disclosed without our express permission. If you are not the intended
> > recipient or an employee or agent responsible for delivering this message
> > to
> > the intended recipient, you are hereby notified that you have received
> > this
> > message in error and that any review, dissemination, distribution or
> > copying
> > of this message, or any attachment thereto, in whole or in part, is
> > strictly
> > prohibited. If you have received this message in error, please immediately
> > notify us by telephone, fax or e-mail and delete the message and all of
> > its
> > attachments.
> > 
> > Thank you.
> > 
> > Every effort is made to keep our network free from viruses. You should,
> > however, review this e-mail message, as well as any attachment thereto,
> > for viruses. We take no responsibility and have no liability for any
> > computer virus which may be transferred via this e-mail message.
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > Join Us At ClueCon - Aug 7-9, 2012
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org


From slava at tangramltd.com  Wed Aug 15 00:36:58 2012
From: slava at tangramltd.com (Dubrovskiy Viacheslav)
Date: Tue, 14 Aug 2012 23:36:58 +0300
Subject: [Freeswitch-users] Any good Canadian SIP carriers
In-Reply-To: <C69FF59885AC464896A4B68500B6E974098CF5E2@VAL-SV-EXCH-01.valiant.com>
References: <C69FF59885AC464896A4B68500B6E974098CF5E2@VAL-SV-EXCH-01.valiant.com>
Message-ID: <502AB6EA.2080601@tangramltd.com>

14.08.2012 23:08, Phil Kim ???????:
>
> Hello all,
>
>  
>
> I am looking for a good SIP carrier (provider) in Canada. We already
> have a carrier in U.S. but they do not offer good rates for calls
> coming over from Canada.
>
>  
>
> Any inputs will be appreciated.
>
We use voip.ms

-- 
WBR,
Dubrovskiy Viacheslav

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From phil.kim at valiant.com  Wed Aug 15 01:19:02 2012
From: phil.kim at valiant.com (Phil Kim)
Date: Tue, 14 Aug 2012 17:19:02 -0400
Subject: [Freeswitch-users] Detecting payphone
Message-ID: <C69FF59885AC464896A4B68500B6E974098CF6BB@VAL-SV-EXCH-01.valiant.com>

Hi all,

 

Question might sounds odd but is there any way to detect if a call is
coming from a 'payphone'?

 

We have an IVR system which people call in to check in/out of there
work. Some calls are coming from a 'payphone'.

 

Problem is our carrier is charging us a huge 'surcharge' on those calls
maid from 'payphone'. We would like to block those calls but not sure
how to detect them.

 

Is there any flag on SIP messages?

 

Thank you in advance.

 

Best regards,

 

Phil H. Kim

Software Engineer

VALIANT Workforce Solutions

phil.kim at valiant.com

www.valiant.com <http://www.valiant.com/> 

516-224-1908

 

The information contained in this e-mail message, and any attachment
thereto, is the property of Valiant and is confidential and may not be
disclosed without our express permission. If you are not the intended
recipient or an employee or agent responsible for delivering this message to
the intended recipient, you are hereby notified that you have received this
message in error and that any review, dissemination, distribution or copying
of this message, or any attachment thereto, in whole or in part, is strictly
prohibited. If you have received this message in error, please immediately
notify us by telephone, fax or e-mail and delete the message and all of its
attachments. 

Thank you. 

Every effort is made to keep our network free from viruses. You should,
however, review this e-mail message, as well as any attachment thereto,
for viruses. We take no responsibility and have no liability for any
computer virus which may be transferred via this e-mail message.
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From sos at sokhapkin.dyndns.org  Wed Aug 15 01:26:47 2012
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 14 Aug 2012 17:26:47 -0400
Subject: [Freeswitch-users] Detecting payphone
In-Reply-To: <C69FF59885AC464896A4B68500B6E974098CF6BB@VAL-SV-EXCH-01.valiant.com>
References: <C69FF59885AC464896A4B68500B6E974098CF6BB@VAL-SV-EXCH-01.valiant.com>
Message-ID: <2373910.jEU7R7Va5k@sos>

 The presence of such kind of information in SIP headers depends on your DID 
provider. Contact your carrier to find the answer.

On Tuesday 14 August 2012 17:19:02 Phil Kim wrote:
> Hi all,
> 
> 
> 
> Question might sounds odd but is there any way to detect if a call is
> coming from a 'payphone'?
> 
> 
> 
> We have an IVR system which people call in to check in/out of there
> work. Some calls are coming from a 'payphone'.
> 
> 
> 
> Problem is our carrier is charging us a huge 'surcharge' on those calls
> maid from 'payphone'. We would like to block those calls but not sure
> how to detect them.
> 
> 
> 
> Is there any flag on SIP messages?
> 
> 
> 
> Thank you in advance.
> 
> 
> 
> Best regards,
> 
> 
> 
> Phil H. Kim
> 
> Software Engineer
> 
> VALIANT Workforce Solutions
> 
> phil.kim at valiant.com
> 
> www.valiant.com <http://www.valiant.com/>
> 
> 516-224-1908
> 
> 
> 
> The information contained in this e-mail message, and any attachment
> thereto, is the property of Valiant and is confidential and may not be
> disclosed without our express permission. If you are not the intended
> recipient or an employee or agent responsible for delivering this message to
> the intended recipient, you are hereby notified that you have received this
> message in error and that any review, dissemination, distribution or
> copying of this message, or any attachment thereto, in whole or in part, is
> strictly prohibited. If you have received this message in error, please
> immediately notify us by telephone, fax or e-mail and delete the message
> and all of its attachments.
> 
> Thank you.
> 
> Every effort is made to keep our network free from viruses. You should,
> however, review this e-mail message, as well as any attachment thereto,
> for viruses. We take no responsibility and have no liability for any
> computer virus which may be transferred via this e-mail message.


From avi at avimarcus.net  Wed Aug 15 01:28:37 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 15 Aug 2012 00:28:37 +0300
Subject: [Freeswitch-users] Detecting payphone
In-Reply-To: <C69FF59885AC464896A4B68500B6E974098CF6BB@VAL-SV-EXCH-01.valiant.com>
References: <C69FF59885AC464896A4B68500B6E974098CF6BB@VAL-SV-EXCH-01.valiant.com>
Message-ID: <CALNZuwnHJopoJpvUhB4G+asdrJbfOyZfXLvAoeoyyoiNO8Go9A@mail.gmail.com>

>From what I know, the carrier has to either offer that option or flag it
for you, so you can reject it
There's no standard for flagging it, nor do all companies, and there's also
no easy way to look up the number in a database to check if it's a payphone.

If someone can tell me otherwise or list TF origination providers that DO
flag it that would be helpful.

-Avi
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From bobc at devassert.com  Wed Aug 15 01:28:45 2012
From: bobc at devassert.com (Bob Coleman)
Date: Tue, 14 Aug 2012 21:28:45 +0000
Subject: [Freeswitch-users] Detecting payphone
In-Reply-To: <C69FF59885AC464896A4B68500B6E974098CF6BB@VAL-SV-EXCH-01.valiant.com>
References: <C69FF59885AC464896A4B68500B6E974098CF6BB@VAL-SV-EXCH-01.valiant.com>
Message-ID: <BAY002-W224305380A363560E898CF4CCB70@phx.gbl>

Hi Phil, We run the same sort of service as you, and have faced the same issue. In some countries they provide you with a list of all payphones, not sure whether the US does, but we found that to be the easiest way to record the surcharge. Bob
 Date: Tue, 14 Aug 2012 17:19:02 -0400
From: phil.kim at valiant.com
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Detecting payphone














Hi all,

 

Question might sounds odd but is there any way to detect if
a call is coming from a ?payphone??

 

We have an IVR system which people call in to check in/out
of there work. Some calls are coming from a ?payphone?.

 

Problem is our carrier is charging us a huge ?surcharge?
on those calls maid from ?payphone?. We would like to block those
calls but not sure how to detect them.

 

Is there any flag on SIP messages?

 

Thank you in advance.

 

Best regards,

 

Phil H. Kim

Software Engineer

VALIANT
Workforce Solutions

phil.kim at valiant.com

www.valiant.com

516-224-1908

 



The information contained in this e-mail message, and any attachment

thereto, is the property of Valiant and is confidential and may not be

disclosed without our express permission. If you are not the intended

recipient or an employee or agent responsible for delivering this message to

the intended recipient, you are hereby notified that you have received this

message in error and that any review, dissemination, distribution or copying

of this message, or any attachment thereto, in whole or in part, is strictly

prohibited. If you have received this message in error, please immediately

notify us by telephone, fax or e-mail and delete the message and all of its

attachments. 

Thank you. 

Every effort is made to keep our network free from viruses. You should,

however, review this e-mail message, as well as any attachment thereto,

for viruses. We take no responsibility and have no liability for any

computer virus which may be transferred via this e-mail message.



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing 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 avi at avimarcus.net  Wed Aug 15 01:36:54 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 15 Aug 2012 00:36:54 +0300
Subject: [Freeswitch-users] Any good Canadian SIP carriers
In-Reply-To: <502AB6EA.2080601@tangramltd.com>
References: <C69FF59885AC464896A4B68500B6E974098CF5E2@VAL-SV-EXCH-01.valiant.com>
	<502AB6EA.2080601@tangramltd.com>
Message-ID: <CALNZuwnoFhNyMFMYMcqd4r+2063RY2wSyMf4yAObWDyqXQpN=Q@mail.gmail.com>

Is there any Canada Orig that's good for a calling card expanding to Canada?

I see prices ~$10/channel, while I have USA/UK/Israel with nearly no
channel fees...

-Avi
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From mike.burlingame at me.com  Wed Aug 15 01:37:32 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Tue, 14 Aug 2012 14:37:32 -0700
Subject: [Freeswitch-users] Detecting payphone
In-Reply-To: <BAY002-W224305380A363560E898CF4CCB70@phx.gbl>
References: <C69FF59885AC464896A4B68500B6E974098CF6BB@VAL-SV-EXCH-01.valiant.com>
	<BAY002-W224305380A363560E898CF4CCB70@phx.gbl>
Message-ID: <5DF02637-7A5A-40E5-9244-8D04811C4201@me.com>

you would want your carrier to send you the isup-oli= field like below, based on the isup-oil= you would make your routing decisions  

 From: <sip:+1XXXXXXXXXX at YYY.YYY.YYY.YYY:5060;isup-oli=27>;tag=gK1e00826b

Ref http://www.nanpa.com/number_resource_info/ani_ii_assignments.html for ANI-II Codes 27 and 70 are generally pay phone people also seem to block 29 prison calls



On Aug 14, 2012, at 2:28 PM, Bob Coleman <bobc at devassert.com> wrote:

> Hi Phil,
>  
> We run the same sort of service as you, and have faced the same issue.
>  
> In some countries they provide you with a list of all payphones, not sure whether the US does, but we found that to be the easiest way to record the surcharge.
>  
> Bob
>  
> Date: Tue, 14 Aug 2012 17:19:02 -0400
> From: phil.kim at valiant.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Detecting payphone
> 
> Hi all,
>  
> Question might sounds odd but is there any way to detect if a call is coming from a ?payphone??
>  
> We have an IVR system which people call in to check in/out of there work. Some calls are coming from a ?payphone?.
>  
> Problem is our carrier is charging us a huge ?surcharge? on those calls maid from ?payphone?. We would like to block those calls but not sure how to detect them.
>  
> Is there any flag on SIP messages?
>  
> Thank you in advance.
>  
> Best regards,
>  
> Phil H. Kim
> Software Engineer
> VALIANT Workforce Solutions
> phil.kim at valiant.com
> www.valiant.com
> 516-224-1908
>  
> The information contained in this e-mail message, and any attachment
> thereto, is the property of Valiant and is confidential and may not be
> disclosed without our express permission. If you are not the intended
> recipient or an employee or agent responsible for delivering this message to
> the intended recipient, you are hereby notified that you have received this
> message in error and that any review, dissemination, distribution or copying
> of this message, or any attachment thereto, in whole or in part, is strictly
> prohibited. If you have received this message in error, please immediately
> notify us by telephone, fax or e-mail and delete the message and all of its
> attachments. 
> Thank you. 
> Every effort is made to keep our network free from viruses. You should,
> however, review this e-mail message, as well as any attachment thereto,
> for viruses. We take no responsibility and have no liability for any
> computer virus which may be transferred via this e-mail message.
> 
> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From phil.kim at valiant.com  Wed Aug 15 01:41:23 2012
From: phil.kim at valiant.com (Phil Kim)
Date: Tue, 14 Aug 2012 17:41:23 -0400
Subject: [Freeswitch-users] Detecting payphone
In-Reply-To: <5DF02637-7A5A-40E5-9244-8D04811C4201@me.com>
References: <C69FF59885AC464896A4B68500B6E974098CF6BB@VAL-SV-EXCH-01.valiant.com><BAY002-W224305380A363560E898CF4CCB70@phx.gbl>
	<5DF02637-7A5A-40E5-9244-8D04811C4201@me.com>
Message-ID: <C69FF59885AC464896A4B68500B6E974098CF6EE@VAL-SV-EXCH-01.valiant.com>

Thank you everyone for help.

 

I will consult it with my provider.

 

 

Best regards,

 

Phil H. Kim

Software Engineer

VALIANT Workforce Solutions

phil.kim at valiant.com

www.valiant.com <http://www.valiant.com/> 

516-224-1908

________________________________

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike
Burlingame
Sent: Tuesday, August 14, 2012 5:38 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Detecting payphone

 

you would want your carrier to send you the isup-oli= field like below,
based on the isup-oil= you would make your routing decisions  

 

 From:
<sip:+1XXXXXXXXXX at YYY.YYY.YYY.YYY:5060;isup-oli=27>;tag=gK1e00826b

 

Ref http://www.nanpa.com/number_resource_info/ani_ii_assignments.html
for ANI-II Codes 27 and 70 are generally pay phone people also seem to
block 29 prison calls

 

 

 

On Aug 14, 2012, at 2:28 PM, Bob Coleman <bobc at devassert.com> wrote:





Hi Phil,
 
We run the same sort of service as you, and have faced the same issue.
 
In some countries they provide you with a list of all payphones, not
sure whether the US does, but we found that to be the easiest way to
record the surcharge.
 
Bob
 

________________________________

Date: Tue, 14 Aug 2012 17:19:02 -0400
From: phil.kim at valiant.com
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Detecting payphone

Hi all,

 

Question might sounds odd but is there any way to detect if a call is
coming from a 'payphone'?

 

We have an IVR system which people call in to check in/out of there
work. Some calls are coming from a 'payphone'.

 

Problem is our carrier is charging us a huge 'surcharge' on those calls
maid from 'payphone'. We would like to block those calls but not sure
how to detect them.

 

Is there any flag on SIP messages?

 

Thank you in advance.

 

Best regards,

 

Phil H. Kim

Software Engineer

VALIANT Workforce Solutions

phil.kim at valiant.com

www.valiant.com <http://www.valiant.com/> 

516-224-1908

 

The information contained in this e-mail message, and any attachment
thereto, is the property of Valiant and is confidential and may not be
disclosed without our express permission. If you are not the intended
recipient or an employee or agent responsible for delivering this
message to
the intended recipient, you are hereby notified that you have received
this
message in error and that any review, dissemination, distribution or
copying
of this message, or any attachment thereto, in whole or in part, is
strictly
prohibited. If you have received this message in error, please
immediately
notify us by telephone, fax or e-mail and delete the message and all of
its
attachments. 
Thank you. 
Every effort is made to keep our network free from viruses. You should,
however, review this e-mail message, as well as any attachment thereto,
for viruses. We take no responsibility and have no liability for any
computer virus which may be transferred via this e-mail message.

________________________________________________________________________
_ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
CudaTel Communication Server  Official FreeSWITCH
Sites http://www.freeswitch.org http://wiki.freeswitch.org
http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012
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________________________________________________________________________
_
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing list
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The information contained in this e-mail message, and any attachment
thereto, is the property of Valiant and is confidential and may not be
disclosed without our express permission. If you are not the intended
recipient or an employee or agent responsible for delivering this message to
the intended recipient, you are hereby notified that you have received this
message in error and that any review, dissemination, distribution or copying
of this message, or any attachment thereto, in whole or in part, is strictly
prohibited. If you have received this message in error, please immediately
notify us by telephone, fax or e-mail and delete the message and all of its
attachments. 

Thank you. 

Every effort is made to keep our network free from viruses. You should,
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computer virus which may be transferred via this e-mail message.
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From bfertig at telstarhosted.com  Wed Aug 15 01:25:49 2012
From: bfertig at telstarhosted.com (Brian Fertig)
Date: Tue, 14 Aug 2012 17:25:49 -0400
Subject: [Freeswitch-users] Detecting payphone
In-Reply-To: <C69FF59885AC464896A4B68500B6E974098CF6BB@VAL-SV-EXCH-01.valiant.com>
References: <C69FF59885AC464896A4B68500B6E974098CF6BB@VAL-SV-EXCH-01.valiant.com>
Message-ID: <171ce4ba08f05bbf4bbab0527a022dc2@mail.gmail.com>

Yes that is fairly simple.  Its called OLI your upstream provider would
pass these digits to you.   From there you can use channel variable to get
the information.



The OLI digits you are looking for are 23 and 27 I believe.







----

Brian Fertig

Director of IT

Telstar Hosted Services

Office: 877.483.5782

Direct: 720.505.4607

Cell: 434.420.5324



[image: Description: Description: cid:image002.png at 01CBDCAF.07A013F0]

*"Your Call Center in the Cloud"*



*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Phil Kim
*Sent:* Tuesday, August 14, 2012 5:19 PM
*To:* FreeSWITCH Users Help
*Subject:* [Freeswitch-users] Detecting payphone



Hi all,



Question might sounds odd but is there any way to detect if a call is
coming from a ?payphone??



We have an IVR system which people call in to check in/out of there work.
Some calls are coming from a ?payphone?.



Problem is our carrier is charging us a huge ?surcharge? on those calls
maid from ?payphone?. We would like to block those calls but not sure how
to detect them.



Is there any flag on SIP messages?



Thank you in advance.



Best regards,



Phil H. Kim

Software Engineer

*VALIANT *Workforce Solutions

phil.kim at valiant.com

www.valiant.com

516-224-1908



The information contained in this e-mail message, and any attachment
thereto, is the property of Valiant and is confidential and may not be
disclosed without our express permission. If you are not the intended
recipient or an employee or agent responsible for delivering this message to
the intended recipient, you are hereby notified that you have received this
message in error and that any review, dissemination, distribution or copying
of this message, or any attachment thereto, in whole or in part, is strictly
prohibited. If you have received this message in error, please immediately
notify us by telephone, fax or e-mail and delete the message and all of its
attachments.

Thank you.

Every effort is made to keep our network free from viruses. You should,
however, review this e-mail message, as well as any attachment thereto,
for viruses. We take no responsibility and have no liability for any
computer virus which may be transferred via this e-mail message.
------------------------------

No virus found in this message.
Checked by AVG - www.avg.com
Version: 2012.0.2197 / Virus Database: 2437/5200 - Release Date: 08/14/12
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From asaad2 at gmail.com  Wed Aug 15 02:12:13 2012
From: asaad2 at gmail.com (BookBag)
Date: Tue, 14 Aug 2012 17:12:13 -0500
Subject: [Freeswitch-users] dumping all variables to freeswitch console or
	logs
Message-ID: <CAAdTMa-=uoN7JSbS3JGKFAD=DU+0nvr6wdM3i+_LeyT_xuMerg@mail.gmail.com>

Hello everyone, I currently have my google voice sending calls to a DID on
my freeswitch server. What I want to do is be able to distinguish between
users who called my DID directly and those who called my google voice
number which forwarded them to my DID.

Unfortunatley for some reason google is sending an empty rdnis and an ani2
(anii). My question is how can I dump all variables to a log or the console
or if anyone knows what variable I can test against that google sends.

Thanx in advance
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From krice at freeswitch.org  Wed Aug 15 02:14:53 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 14 Aug 2012 17:14:53 -0500
Subject: [Freeswitch-users] dumping all variables to freeswitch console
 or logs
In-Reply-To: <CAAdTMa-=uoN7JSbS3JGKFAD=DU+0nvr6wdM3i+_LeyT_xuMerg@mail.gmail.com>
Message-ID: <CC50380D.3C052%krice@freeswitch.org>

Info application in your dialplan... If you are using the example configs as
a starting point look for the debug extension

Also if you have a call up you want to look at all the channel variables
from the fs cli

Show channels, get the UUID of the incoming let, then uuid_dump UUID will
output all the goodness to the console

K


On 8/14/12 5:12 PM, "BookBag" <asaad2 at gmail.com> wrote:

> Hello everyone, I currently have my google voice sending calls to a DID on my
> freeswitch server. What I want to do is be able to distinguish between users
> who called my DID directly and those who called my google voice number which
> forwarded them to my DID.
> 
> Unfortunatley for some reason google is sending an empty rdnis and an ani2
> (anii). My question is how can I dump all variables to a log or the console or
> if anyone knows what variable I can test against that google sends.
> 
> Thanx in advance
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 darknesslabs at gmail.com  Wed Aug 15 02:20:41 2012
From: darknesslabs at gmail.com (Karol)
Date: Tue, 14 Aug 2012 18:20:41 -0400
Subject: [Freeswitch-users] Detecting payphone
In-Reply-To: <171ce4ba08f05bbf4bbab0527a022dc2@mail.gmail.com>
References: <C69FF59885AC464896A4B68500B6E974098CF6BB@VAL-SV-EXCH-01.valiant.com>
	<171ce4ba08f05bbf4bbab0527a022dc2@mail.gmail.com>
Message-ID: <CAFZs+v_hBEuKess9C=j3rK0uvU_FzYNgkRnqNu8r6SYONVPSAQ@mail.gmail.com>

http://www.dialogic.com/webhelp/img1010/10.5.1/webhelp/oli_isdn_sip.htm

Just found that URL. Should help fine tune the codes you are looking for.
On Aug 14, 2012 6:13 PM, "Brian Fertig" <bfertig at telstarhosted.com> wrote:

> Yes that is fairly simple.  Its called OLI your upstream provider would
> pass these digits to you.   From there you can use channel variable to get
> the information.
>
>
>
> The OLI digits you are looking for are 23 and 27 I believe.
>
>
>
>
>
>
>
> ----
>
> Brian Fertig
>
> Director of IT
>
> Telstar Hosted Services
>
> Office: 877.483.5782
>
> Direct: 720.505.4607
>
> Cell: 434.420.5324
>
>
>
> [image: Description: Description: cid:image002.png at 01CBDCAF.07A013F0]
>
> *"Your Call Center in the Cloud"*
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Phil Kim
> *Sent:* Tuesday, August 14, 2012 5:19 PM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Detecting payphone
>
>
>
> Hi all,
>
>
>
> Question might sounds odd but is there any way to detect if a call is
> coming from a ?payphone??
>
>
>
> We have an IVR system which people call in to check in/out of there work.
> Some calls are coming from a ?payphone?.
>
>
>
> Problem is our carrier is charging us a huge ?surcharge? on those calls
> maid from ?payphone?. We would like to block those calls but not sure how
> to detect them.
>
>
>
> Is there any flag on SIP messages?
>
>
>
> Thank you in advance.
>
>
>
> Best regards,
>
>
>
> Phil H. Kim
>
> Software Engineer
>
> *VALIANT *Workforce Solutions
>
> phil.kim at valiant.com
>
> www.valiant.com
>
> 516-224-1908
>
>
>
> The information contained in this e-mail message, and any attachment
> thereto, is the property of Valiant and is confidential and may not be
> disclosed without our express permission. If you are not the intended
> recipient or an employee or agent responsible for delivering this message
> to
> the intended recipient, you are hereby notified that you have received this
> message in error and that any review, dissemination, distribution or
> copying
> of this message, or any attachment thereto, in whole or in part, is
> strictly
> prohibited. If you have received this message in error, please immediately
> notify us by telephone, fax or e-mail and delete the message and all of its
> attachments.
>
> Thank you.
>
> Every effort is made to keep our network free from viruses. You should,
> however, review this e-mail message, as well as any attachment thereto,
> for viruses. We take no responsibility and have no liability for any
> computer virus which may be transferred via this e-mail message.
> ------------------------------
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2197 / Virus Database: 2437/5200 - Release Date: 08/14/12
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 spencer at 5ninesolutions.com  Wed Aug 15 02:25:58 2012
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Tue, 14 Aug 2012 15:25:58 -0700
Subject: [Freeswitch-users] SUBSCRIBE for MWI
Message-ID: <81887D6C-EB8F-4B42-9FFD-6CC46F768D7D@5ninesolutions.com>

Hello,
I'm trying to use SUBSCRIBEs to monitor a shared VM box.  I'm only receiving NOTIFYs after the number of voicemail messages changes, not when the subscription is created.  I'm on latest git revision built this morning.  Is there any option that must be enabled to send a NOTIFY when the subscription is created?

Thanks, Spencer



   ------------------------------------------------------------------------
recv 450 bytes from udp/[x.x.x.x]:5061 at 22:10:13.122762:
   ------------------------------------------------------------------------
   SUBSCRIBE sip:1000 at pbx03.domain.com SIP/2.0
   Via: SIP/2.0/UDP 10.59.1.249:5061;branch=z9hG4bK-e18f92db;rport
   From: "Test" <sip:1007 at pbx03.domain.com>;tag=2d525326a41799c3
   To: <sip:1000 at pbx03.domain.com>
   Call-ID: 722b1d22-bc56f7f7 at 10.59.1.249
   CSeq: 35007 SUBSCRIBE
   Max-Forwards: 70
   Contact: "Test" <sip:1007 at 10.59.1.249:5061>
   Expires: 3600
   Event: message-summary
   User-Agent: Cisco/SPA509G-7.5.2b
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 773 bytes to udp/[x.x.x.x]:5061 at 22:10:13.124509:
   ------------------------------------------------------------------------
   SIP/2.0 202 Accepted
   Via: SIP/2.0/UDP 10.59.1.249:5061;branch=z9hG4bK-e18f92db;rport=5061;received=x.x.x.x
   From: "Test" <sip:1007 at pbx03.domain.com>;tag=2d525326a41799c3
   To: <sip:1000 at pbx03.domain.com>;tag=3ZTmO9g9lhPo
   Call-ID: 722b1d22-bc56f7f7 at 10.59.1.249
   CSeq: 35007 SUBSCRIBE
   Contact: <sip:1000 at x.x.x.x:5070>
   Expires: 3600
   User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: precondition, path, replaces
   Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Subscription-State: active;expires=3600
   Content-Length: 0



No NOTIFY is sent until after changing the number of voicemail messages:
   
send 1018 bytes to udp/[x.x.x.x]:5061 at 22:14:50.879672:
   ------------------------------------------------------------------------
   NOTIFY sip:1007 at 10.59.1.249:5061 SIP/2.0
   Via: SIP/2.0/UDP x.x.x.x:5070;rport;branch=z9hG4bKa4Qc71DKS83pm
   Route: <sip:x.x.x.x:5061>;transport=udp
   Max-Forwards: 70
   From: <sip:1000 at pbx03.domain.com>;tag=3ZTmO9g9lhPo
   To: "Test" <sip:1007 at pbx03.domain.com>;tag=2d525326a41799c3
   Call-ID: 722b1d22-bc56f7f7 at 10.59.1.249
   CSeq: 322887251 NOTIFY
   Contact: <sip:1000 at x.x.x.x:5070;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: precondition, path, replaces
   Event: message-summary
   Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Subscription-State: active;expires=3323
   Content-Type: application/simple-message-summary
   Content-Length: 105
   
   Messages-Waiting: yes
   Message-Account: sip:1000 at pbx03.domain.com
   Voice-Message: 148/0 (0/0)



From asaad2 at gmail.com  Wed Aug 15 02:26:06 2012
From: asaad2 at gmail.com (BookBag)
Date: Tue, 14 Aug 2012 17:26:06 -0500
Subject: [Freeswitch-users] dumping all variables to freeswitch console
	or logs
In-Reply-To: <CC50380D.3C052%krice@freeswitch.org>
References: <CAAdTMa-=uoN7JSbS3JGKFAD=DU+0nvr6wdM3i+_LeyT_xuMerg@mail.gmail.com>
	<CC50380D.3C052%krice@freeswitch.org>
Message-ID: <CAAdTMa_HV7SQTarD07AmmrJXbq1KySOYOc1gaLdn+Tffnk7OJQ@mail.gmail.com>

Thanks ken, I think I'll try to set a call up and do that. I was also
reading about verbose_events (true). But I'm confused how to use it. Is it
possible to use that to dump to console.
On Aug 14, 2012 6:17 PM, "Ken Rice" <krice at freeswitch.org> wrote:

>  Info application in your dialplan... If you are using the example
> configs as a starting point look for the debug extension
>
> Also if you have a call up you want to look at all the channel variables
> from the fs cli
>
> Show channels, get the UUID of the incoming let, then uuid_dump UUID will
> output all the goodness to the console
>
> K
>
>
> On 8/14/12 5:12 PM, "BookBag" <asaad2 at gmail.com> wrote:
>
> Hello everyone, I currently have my google voice sending calls to a DID on
> my freeswitch server. What I want to do is be able to distinguish between
> users who called my DID directly and those who called my google voice
> number which forwarded them to my DID.
>
> Unfortunatley for some reason google is sending an empty rdnis and an ani2
> (anii). My question is how can I dump all variables to a log or the console
> or if anyone knows what variable I can test against that google sends.
>
> Thanx in advance
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 asaad2 at gmail.com  Wed Aug 15 03:06:57 2012
From: asaad2 at gmail.com (BookBag)
Date: Tue, 14 Aug 2012 18:06:57 -0500
Subject: [Freeswitch-users] dumping all variables to freeswitch console
	or logs
In-Reply-To: <CAAdTMa_HV7SQTarD07AmmrJXbq1KySOYOc1gaLdn+Tffnk7OJQ@mail.gmail.com>
References: <CAAdTMa-=uoN7JSbS3JGKFAD=DU+0nvr6wdM3i+_LeyT_xuMerg@mail.gmail.com>
	<CC50380D.3C052%krice@freeswitch.org>
	<CAAdTMa_HV7SQTarD07AmmrJXbq1KySOYOc1gaLdn+Tffnk7OJQ@mail.gmail.com>
Message-ID: <CAAdTMa8uUmf-rXgPA539Dk7buZt0sDXUws1m1fy+JDAJxLZHcg@mail.gmail.com>

Thanx ken, application info worked like a charm
On Aug 14, 2012 6:26 PM, "BookBag" <asaad2 at gmail.com> wrote:

> Thanks ken, I think I'll try to set a call up and do that. I was also
> reading about verbose_events (true). But I'm confused how to use it. Is it
> possible to use that to dump to console.
> On Aug 14, 2012 6:17 PM, "Ken Rice" <krice at freeswitch.org> wrote:
>
>>  Info application in your dialplan... If you are using the example
>> configs as a starting point look for the debug extension
>>
>> Also if you have a call up you want to look at all the channel variables
>> from the fs cli
>>
>> Show channels, get the UUID of the incoming let, then uuid_dump UUID will
>> output all the goodness to the console
>>
>> K
>>
>>
>> On 8/14/12 5:12 PM, "BookBag" <asaad2 at gmail.com> wrote:
>>
>> Hello everyone, I currently have my google voice sending calls to a DID
>> on my freeswitch server. What I want to do is be able to distinguish
>> between users who called my DID directly and those who called my google
>> voice number which forwarded them to my DID.
>>
>> Unfortunatley for some reason google is sending an empty rdnis and an
>> ani2 (anii). My question is how can I dump all variables to a log or the
>> console or if anyone knows what variable I can test against that google
>> sends.
>>
>> Thanx in advance
>>
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 gabe at gundy.org  Wed Aug 15 03:21:05 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 14 Aug 2012 17:21:05 -0600
Subject: [Freeswitch-users] FreeSWITCH 1.2.1 has been rolled...
In-Reply-To: <CC4F64C1.3C00B%krice@freeswitch.org>
References: <CC4F64C1.3C00B%krice@freeswitch.org>
Message-ID: <CAMwZ-tLwKT-X-d2SD9HhhWCEX49+5hfQOGgEEcVYxXyLfqx4Ew@mail.gmail.com>

On Tue, Aug 14, 2012 at 1:13 AM, Ken Rice <krice at freeswitch.org> wrote:
> Also Debian users, check out the debian repo on
> http://files.freeswitch.org/repo  More info on this coming soon, currently
> this continues nightly builds from the dev branch but will soon also contain
> a stable repo...

I'm glad to see some debian love :)


Gabe


From curriegrad2004 at gmail.com  Wed Aug 15 03:30:37 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 14 Aug 2012 16:30:37 -0700
Subject: [Freeswitch-users] Any good Canadian SIP carriers
In-Reply-To: <CALNZuwnoFhNyMFMYMcqd4r+2063RY2wSyMf4yAObWDyqXQpN=Q@mail.gmail.com>
References: <C69FF59885AC464896A4B68500B6E974098CF5E2@VAL-SV-EXCH-01.valiant.com>
	<502AB6EA.2080601@tangramltd.com>
	<CALNZuwnoFhNyMFMYMcqd4r+2063RY2wSyMf4yAObWDyqXQpN=Q@mail.gmail.com>
Message-ID: <CAMWiMy8kjv+WzUuKgR_GggGy9Z50UThWdsb_rUgetPW6P=9AHA@mail.gmail.com>

Try voicenetwork.ca for canadian term

On Tue, Aug 14, 2012 at 2:36 PM, Avi Marcus <avi at avimarcus.net> wrote:
> Is there any Canada Orig that's good for a calling card expanding to Canada?
>
> I see prices ~$10/channel, while I have USA/UK/Israel with nearly no channel
> fees...
>
> -Avi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From Adam.Lappe at qsc.de  Wed Aug 15 08:33:53 2012
From: Adam.Lappe at qsc.de (Lappe, Adam)
Date: Wed, 15 Aug 2012 06:33:53 +0200
Subject: [Freeswitch-users] sendevent to Gateway
In-Reply-To: <CAFGRPVqby1X4JxtEFgKeGEre6yRugsvsOFND4JSoUbfMJSDRKQ@mail.gmail.com>
References: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EBF@CGNMSG01.oc.qsc.de>
	<F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EC1@CGNMSG01.oc.qsc.de>
	<CAFGRPVro3EE2ug57k4bkmZHPGbF1EcquQeN_vtQ08dKs-nAxMQ@mail.gmail.com>
	<F1EBD7E7528EA446BB9898D22914AA3B3D8FC57EC4@CGNMSG01.oc.qsc.de>
	<CAFGRPVqby1X4JxtEFgKeGEre6yRugsvsOFND4JSoUbfMJSDRKQ@mail.gmail.com>
Message-ID: <F1EBD7E7528EA446BB9898D22914AA3B3D8FC57ED2@CGNMSG01.oc.qsc.de>

Hello David,

sorry for the late response.

Thank you for the example. Yes, I tried it already.
But the problem is this part:


user: 1005

host: 192.168.10.4

I am trying to send this Message to my gateway.


sofia status:
                    Name        Type                                       Date State
=================================================================================================
                 external    profile             sip:mod_sofia at ip-Address:Port   RUNNING (0)
        external::gateway    gateway             sip:login at realm                REGED

________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von David Villasmil
Gesendet: Donnerstag, 9. August 2012 14:07
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] sendevent to Gateway

Have you tried:

http://wiki.freeswitch.org/wiki/Event_Socket#sendevent


Example of sendevent with a message body, the length of the body is specified by content-length:

sendevent NOTIFY

profile: internal

content-type: application/simple-message-summary

event-string: check-sync

user: 1005

host: 192.168.10.4

content-length: 2



OK

Another example with a notify:





sendevent NOTIFY

profile: internal

content-type: application/simple-message-summary

event-string: check-sync

user: 1005

host: 99.157.44.194

content-length: 2



OK

Results in a packet like this:





NOTIFY sip:1005 at 99.157.44.203<mailto:sip%3A1005 at 99.157.44.203> SIP/2.0

Via: SIP/2.0/UDP 99.157.44.194;rport;branch=z9hG4bKpH2DtBDcDtg0N

Max-Forwards: 70

From: <sip:1005 at 99.157.44.194<mailto:sip%3A1005 at 99.157.44.194>>;tag=Dy3c6Q1y15v5S

To: <sip:1005 at 99.157.44.194<mailto:sip%3A1005 at 99.157.44.194>>

Call-ID: 129d1446-0063-122c-15aa-001a923f6a0f

CSeq: 104766492 NOTIFY

Contact: <sip:mod_sofia at 99.157.44.194:5060<http://sip:mod_sofia at 99.157.44.194:5060>>

User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9578:9586

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH

Supported: 100rel, timer, precondition, path, replaces

Event: check-sync

Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary

Subscription-State: terminated;timeout

Content-Type: application/simple-message-summary

Content-Length: 2



OK

On Wed, Aug 8, 2012 at 10:40 AM, Lappe, Adam <Adam.Lappe at qsc.de<mailto:Adam.Lappe at qsc.de>> wrote:
I am trying to send a NOTIFY Message to my Gateway that looks like this:

NOTIFY sip:(address)@(host):(port) SIP/2.0
VIA: SIP/2.0/UDP (address):(port)
From:;tag=1
To:
Call-ID: whatever
CSeq: 1 NOTIFY
Contact:
Event: message-summary
Subscription-State: terminated
Content-Type: application/simple-message-summary
Content-Length: 23

Messages-Waiting: yes


Is it possible to make the FreeSWITCH send such a message to my gateway?

Thanks,
Adam
________________________________
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 David Villasmil
Gesendet: Dienstag, 7. August 2012 11:59
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] sendevent to Gateway

Hello,

What EXCATLY do you want to send?

David
On Tue, Aug 7, 2012 at 11:48 AM, Lappe, Adam <Adam.Lappe at qsc.de<mailto:Adam.Lappe at qsc.de>> wrote:
I am still looking for a solution.
Is there no one who has an idea?

Thanks,
Adam

________________________________
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 Lappe, Adam
Gesendet: Freitag, 3. August 2012 11:53
An: 'FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>'
Betreff: [Freeswitch-users] sendevent to Gateway

Hi all,

i am trying to make the freeswitch send a event (or SIP Message) to my gateway.
I can only find examples how to send events to registered endpoint but not to my gateway.

Shouldn?t it be possible to do this via the event socket?

Thanks in advance and best regards,
Adam

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-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 basit.engg at gmail.com  Wed Aug 15 09:52:23 2012
From: basit.engg at gmail.com (Abdul Basit)
Date: Wed, 15 Aug 2012 10:52:23 +0500
Subject: [Freeswitch-users] FreeSWITCH 1.2.1 has been rolled...
In-Reply-To: <CAMWiMy-md2CT3dPbBPa2cQcSG09hiCyo19PBdiN7H6r_J5e1kA@mail.gmail.com>
References: <CC4F64C1.3C00B%krice@freeswitch.org>
	<CAMWiMy-md2CT3dPbBPa2cQcSG09hiCyo19PBdiN7H6r_J5e1kA@mail.gmail.com>
Message-ID: <CAGQG5MYs290WiNLyZVajgTUrPT6vZGu5kCJePv2UfSF5hB9Ovw@mail.gmail.com>

No need to follow microsoft :)

1.2.1 is good enough.


On Tue, Aug 14, 2012 at 12:20 PM, curriegrad2004
<curriegrad2004 at gmail.com>wrote:

> 1.2.1 Should be known as 1.2.0 SP1 :P
>
> On Tue, Aug 14, 2012 at 12:13 AM, Ken Rice <krice at freeswitch.org> wrote:
> > FreeSWITCH 1.2.1 is Here!!!
> >
> > Tarball is on files.freeswitch.org as usual.
> >
> > Also Debian users, check out the debian repo on
> > http://files.freeswitch.org/repo  More info on this coming soon,
> currently
> > this continues nightly builds from the dev branch but will soon also
> contain
> > a stable repo...
> >
> > As usual, any issues please report them via jira.freeswitch.org and
> please
> > tag them with the correct version!
> >
> >
> > K
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jh.zhou at outlook.com  Wed Aug 15 04:06:53 2012
From: jh.zhou at outlook.com (ZhouJianhua)
Date: Wed, 15 Aug 2012 00:06:53 +0000
Subject: [Freeswitch-users] No audio with two PSTN leg
Message-ID: <COL002-W92114B15545FE508BD1E5BF8B60@phx.gbl>

Hi,
I tried to bridge two PSTN call like this:
  originate sofia/gateway/mygateway/num1 &bridge(sofia/gateway/mygateway/num2)
Two phone ringed but no audio. There is audio if one leg is SIP user, like this:
  originate sofia/internal/1000 &bridge(sofia/gateway/mygateway/num2)
or
   originate sofia/gateway/mygateway/num1 &bridge(sofia/internal/1000) 		 	   		  
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From asaad2 at gmail.com  Wed Aug 15 15:31:59 2012
From: asaad2 at gmail.com (BookBag)
Date: Wed, 15 Aug 2012 06:31:59 -0500
Subject: [Freeswitch-users] No audio with two PSTN leg
In-Reply-To: <COL002-W92114B15545FE508BD1E5BF8B60@phx.gbl>
References: <COL002-W92114B15545FE508BD1E5BF8B60@phx.gbl>
Message-ID: <CAAdTMa86HDfzM-KKBR7v3Ey35WL1-mN8wSbodh10RDtNZZgUuQ@mail.gmail.com>

How many channels is your provider allowing you? I remember I had the same
problem with asterisk a long time ago and it turned out my provider only
allowed 2 channels at a time
On Aug 15, 2012 2:21 AM, "ZhouJianhua" <jh.zhou at outlook.com> wrote:

> Hi,
>
> I tried to bridge two PSTN call like this:
>
>   originate sofia/gateway/mygateway/num1 &bridge(
> sofia/gateway/mygateway/num2)
>
> Two phone ringed but no audio. There is audio if one leg is SIP user, like
> this:
>
>   originate sofia/internal/1000 &bridge(sofia/gateway/mygateway/num2)
>
> or
>
>    originate sofia/gateway/mygateway/num1 &bridge(sofia/internal/1000)
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 x.liu at hw.ac.uk  Wed Aug 15 15:34:57 2012
From: x.liu at hw.ac.uk (x.liu)
Date: Wed, 15 Aug 2012 12:34:57 +0100
Subject: [Freeswitch-users] mod_rtmp audio codecs issues for speech
	recognition?
In-Reply-To: <2C3B8DC9-5E1F-4497-947F-B653A47E60A8@freeswitch.org>
References: <1FFF97C269757C458224B7C895F35F15141094@cantor.std.visionutv.se>
	<50236299.4070403@softnet.si>
	<CAKV-jFyJEGW72UwZ8AGkKPL9uvoab3fOUb7y5_wTx6_suPT0JQ@mail.gmail.com>
	<5026C7D8.80008@quentustech.com>
	<Pine.LNX.4.64.1208130308060.16739@1a-8140.antagus.de>
	<1676.1344828871@ccs.covici.com>
	<2C3B8DC9-5E1F-4497-947F-B653A47E60A8@freeswitch.org>
Message-ID: <502B8961.4040703@hw.ac.uk>

Hello,

I am trying the FS sample Flex client for TTS and speech recognition.
The TTS sounds OK. I use Nuance MRCP v1 server with Nuance recognizer 9.

The problem is that I hardly get the recognition result,only 
occasionally I got some simple words output like "repeat", "ok" which 
were not I said.
The whole recognition setup works well when I use a telephone line but 
basically not work by using the flex client from a web broswer.
The recorded voices in FS which were sent from the web browser via the 
flash rtmp sound fine.

I guess it may be the audio codec issues. The flash rtmp uses Speex. My 
mrcp profile specifies the codecs as "PCMU PCMA L16/96/8000"

Is there a way to automatically convert the Speex audio to the mrcp 
required formats during calling the system?
Or what should I look at to get the proper speech recognition via the 
flash rtmp?

Many thanks!

Xing



-- 
Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012

We invite research leaders and ambitious early career researchers to 
join us in leading and driving research in key inter-disciplinary themes. 
Please see www.hw.ac.uk/researchleaders for further information and how
to apply.

Heriot-Watt University is a Scottish charity
registered under charity number SC000278.



From peter.olsson at visionutveckling.se  Wed Aug 15 15:58:30 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 15 Aug 2012 11:58:30 +0000
Subject: [Freeswitch-users] mod_rtmp audio codecs issues for
	speech	recognition?
Message-ID: <1FFF97C269757C458224B7C895F35F151454FA@cantor.std.visionutv.se>

I noticed this as well when I was playing around with UniMRCP and Nuance some time ago. mod_unimrcp  seems to be missing some code to resample the audio frames to the bitrate used by the nuance connection. If I remember correctly it was ok to use (for instance) PCMA on the call leg and PCMU on the nuance leg (they both use 8khz), but as soon as I tried to use a codec with another rate things seemed to stop working.

I never had the time to chase this down any further, and I was only trying it for an internal test case, so it wasn't so important for me.

I believe the best thing you can do is to report this to Jira, and hopefully someone will be able to look into this.

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r x.liu
Skickat: den 15 augusti 2012 13:35
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] mod_rtmp audio codecs issues for speech recognition?

Hello,

I am trying the FS sample Flex client for TTS and speech recognition.
The TTS sounds OK. I use Nuance MRCP v1 server with Nuance recognizer 9.

The problem is that I hardly get the recognition result,only occasionally I got some simple words output like "repeat", "ok" which were not I said.
The whole recognition setup works well when I use a telephone line but basically not work by using the flex client from a web broswer.
The recorded voices in FS which were sent from the web browser via the flash rtmp sound fine.

I guess it may be the audio codec issues. The flash rtmp uses Speex. My mrcp profile specifies the codecs as "PCMU PCMA L16/96/8000"

Is there a way to automatically convert the Speex audio to the mrcp required formats during calling the system?
Or what should I look at to get the proper speech recognition via the flash rtmp?

Many thanks!

Xing



--
Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012

We invite research leaders and ambitious early career researchers to join us in leading and driving research in key inter-disciplinary themes. 
Please see www.hw.ac.uk/researchleaders for further information and how to apply.

Heriot-Watt University is a Scottish charity registered under charity number SC000278.


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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

!DSPAM:502b86f932762019970451!



From steveayre at gmail.com  Wed Aug 15 17:09:35 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 15 Aug 2012 14:09:35 +0100
Subject: [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAKzWOxWCr5X1547qOwj7Zxvg3JB2KDXrJW9zZxntjb-_e292+g@mail.gmail.com>
References: <1344419198.28524.140661112182326.206F22F6@webmail.messagingengine.com>
	<1344428263.15547.140661112231926.2E36F8FB@webmail.messagingengine.com>
	<CAKzWOxWCr5X1547qOwj7Zxvg3JB2KDXrJW9zZxntjb-_e292+g@mail.gmail.com>
Message-ID: <CAFiqYu=OQsPFi+JfwHnPN96UdoK_MNQV2PmhJS3m+zccg+7dGg@mail.gmail.com>

That section is out of date.

There's a debian/ which you can use to build debian packages for
Debian/Ubuntu... someone will need to update it at some point

Assuming you checkout into a freeswitch directory, this should work:
$ cd freeswitch
$ (cd debian && ./bootstrap.sh -c squeeze)
$ dpkg-buildpackage -b

There'll be a ton of .deb files in the parent directory, which you can
install with "dpkg -i filename.deb"
Stick all the ones you need in a single command and it'll keep the
dependancies between fs packages happy.

-Steve




On 8 August 2012 19:22, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 8, 2012 at 5:17 AM, Eelco <eelco at fastmail.nl> wrote:
>
>>   I see a lot of scripts inside .. but what is the correct way to
>> install this tarball on Debian Squeeze ?
>>
>>
>
> If this is a simple Linux install (which it sounds like) then start with
> these instructions:
> http://wiki.freeswitch.org/wiki/Installation_Guide#Debian
>
> -MC
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From x.liu at hw.ac.uk  Wed Aug 15 17:41:18 2012
From: x.liu at hw.ac.uk (x.liu)
Date: Wed, 15 Aug 2012 14:41:18 +0100
Subject: [Freeswitch-users] mod_rtmp audio codecs issues for
	speech	recognition?
In-Reply-To: <1FFF97C269757C458224B7C895F35F151454FA@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F151454FA@cantor.std.visionutv.se>
Message-ID: <502BA6FE.4060705@hw.ac.uk>

Hi Peter,

Many thanks for your response and suggestion!

I will report it to Jira and report back here when it is resolved.

Thanks again,

Xing


On 08/15/2012 12:58 PM, Peter Olsson wrote:
> I noticed this as well when I was playing around with UniMRCP and Nuance some time ago. mod_unimrcp  seems to be missing some code to resample the audio frames to the bitrate used by the nuance connection. If I remember correctly it was ok to use (for instance) PCMA on the call leg and PCMU on the nuance leg (they both use 8khz), but as soon as I tried to use a codec with another rate things seemed to stop working.
>
> I never had the time to chase this down any further, and I was only trying it for an internal test case, so it wasn't so important for me.
>
> I believe the best thing you can do is to report this to Jira, and hopefully someone will be able to look into this.
>
> /Peter
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r x.liu
> Skickat: den 15 augusti 2012 13:35
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] mod_rtmp audio codecs issues for speech recognition?
>
> Hello,
>
> I am trying the FS sample Flex client for TTS and speech recognition.
> The TTS sounds OK. I use Nuance MRCP v1 server with Nuance recognizer 9.
>
> The problem is that I hardly get the recognition result,only occasionally I got some simple words output like "repeat", "ok" which were not I said.
> The whole recognition setup works well when I use a telephone line but basically not work by using the flex client from a web broswer.
> The recorded voices in FS which were sent from the web browser via the flash rtmp sound fine.
>
> I guess it may be the audio codec issues. The flash rtmp uses Speex. My mrcp profile specifies the codecs as "PCMU PCMA L16/96/8000"
>
> Is there a way to automatically convert the Speex audio to the mrcp required formats during calling the system?
> Or what should I look at to get the proper speech recognition via the flash rtmp?
>
> Many thanks!
>
> Xing
>
>
>
> --
> Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012
>
> We invite research leaders and ambitious early career researchers to join us in leading and driving research in key inter-disciplinary themes.
> Please see www.hw.ac.uk/researchleaders for further information and how to apply.
>
> Heriot-Watt University is a Scottish charity registered under charity number SC000278.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
> !DSPAM:502b86f932762019970451!
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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



-- 
Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012

We invite research leaders and ambitious early career researchers to 
join us in leading and driving research in key inter-disciplinary themes. 
Please see www.hw.ac.uk/researchleaders for further information and how
to apply.

Heriot-Watt University is a Scottish charity
registered under charity number SC000278.



From krice at freeswitch.org  Wed Aug 15 19:31:31 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 15 Aug 2012 10:31:31 -0500
Subject: [Freeswitch-users] Weekly FreeSWITCH Conference Call
Message-ID: <CC512B03.3C07B%krice@freeswitch.org>

Hello Gang!

This weeks conf call will be starting soon...

The Agenda Page for today:
http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_15

I will be talking about FreeSWITCH 1.2, how to get it and whats news.

We?ll follow this up with community discussion!

Join us!

Ken
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From info at pripojtese.net  Wed Aug 15 15:22:34 2012
From: info at pripojtese.net (Jakub Tencl)
Date: Wed, 15 Aug 2012 12:22:34 +0100
Subject: [Freeswitch-users] IP authentification
Message-ID: <502B867A.5060608@pripojtese.net>

Hi,

i'm in the middle of the testing our fresh installation and now i can't 
find out howto set IP authentification, when freeswitch are registered 
on the GW, it works fine but without registration, just based on IP i 
have no idea how the freeswitch is doing that if so. Can you help me 
please? I am using the vBilling system.

Thanks

Jakub




From info at pripojtese.net  Wed Aug 15 15:43:35 2012
From: info at pripojtese.net (Jakub Tencl)
Date: Wed, 15 Aug 2012 12:43:35 +0100
Subject: [Freeswitch-users] Limitation
Message-ID: <502B8B67.3090306@pripojtese.net>

Hello,

i'm just wondering what is the limitation in freeswitch and vBilling 
system, i've noticed in the log that i can make 250 concurrent calls and 
50k calls per day and if i am using authentification via IP, is there 
another limitation? Here is log from my testing call:

------------------------------------------------------------------------
send 321 bytes to udp/[x.x.x.x]:5060 at 11:13:48.500060:
------------------------------------------------------------------------
    ACK sip:phone number at x.x.x.x SIP/2.0
    Via: SIP/2.0/UDP x.x.x.x;rport;branch=z9hG4bKXv02SBvK3U8SS
    Max-Forwards: 69
    From: "281545" <sip:281545 at x.x.x.x>;tag=DDNBjUggvFm6B
    To: <sip:phone number at x.x.x.x>
    Call-ID: 20a033f8-616d-1230-0e88-00188b8cb036
    CSeq: 32178550 ACK
    Content-Length: 0

------------------------------------------------------------------------
2012-08-15 12:13:48.496981 [NOTICE] sofia.c:6847 Hangup 
sofia/Billing205/phone number [CS_CONSUME_MEDIA] [CALL_REJECTED]
2012-08-15 12:13:48.496981 [NOTICE] switch_core_session.c:1447 Session 
146 (sofia/Billing205/phone number) Ended
2012-08-15 12:13:48.496981 [INFO] mod_dptools.c:3027 Originate Failed.  
Cause: CALL_REJECTED
2012-08-15 12:13:48.496981 [NOTICE] switch_core_session.c:1449 Close 
Channel sofia/Billing205/phone number [CS_DESTROY]
2012-08-15 12:13:48.496981 [NOTICE] mod_dptools.c:3147 Hangup 
sofia/Billing205/281545 at x.x.x.x:5060 [CS_EXECUTE] [CALL_REJECTED]
2012-08-15 12:13:48.496981 [INFO] mod_hash.c:304 Usage for 
x.x.x.x_192.168.2.96 is now 0
2012-08-15 12:13:48.496981 [INFO] mod_hash.c:304 Usage for 
MAX_CALLS_PER_DAY_IN_FREE_MODE_call_per_day is now 0
2012-08-15 12:13:48.496981 [INFO] mod_hash.c:304 Usage for 
CONCURRENT_CALLS_IN_FREE_MODE_calls_max is now 0
send 752 bytes to udp/[192.168.2.96]:54205 at 11:13:48.502357:
------------------------------------------------------------------------
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 
192.168.5.110:54205;branch=z9hG4bK-d8754z-4b7558087e1e491d-1---d8754z-;rport=54205;received=192.168.2.96
    Max-Forwards: 70
    From: "281545"<sip:281545 at x.x.x.x:5060>;tag=db146842
    To: <sip:phone number at x.x.x.x:5060>;tag=c4Ujg0ZcZ6XKg
    Call-ID: ZDM4YTU2ZmU3ZjNhNTc2MzAyZGZiYmNmMmVjYTRiNzE.
    CSeq: 2 INVITE
    User-Agent: vBilling
    Accept: application/sdp
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
REGISTER, NOTIFY
    Supported: precondition, path, replaces
    Allow-Events: talk, hold, conference, refer
    Reason: Q.850;cause=21;text="CALL_REJECTED"
    Content-Length: 0
    Remote-Party-ID: "phone number" <sip:phone 
number at x.x.x.x>;party=calling;privacy=off;screen=no

------------------------------------------------------------------------
2012-08-15 12:13:48.496981 [NOTICE] switch_core_session.c:1447 Session 
145 (sofia/Billing205/281545 at x.x.x.x:5060) Ended
2012-08-15 12:13:48.496981 [NOTICE] switch_core_session.c:1449 Close 
Channel sofia/Billing205/281545 at x.x.x.x:5060 [CS_DESTROY]

Thanks

Jakub




From jh.zhou at outlook.com  Wed Aug 15 17:10:44 2012
From: jh.zhou at outlook.com (ZhouJianhua)
Date: Wed, 15 Aug 2012 13:10:44 +0000
Subject: [Freeswitch-users] No audio with two PSTN leg
In-Reply-To: <CAAdTMa86HDfzM-KKBR7v3Ey35WL1-mN8wSbodh10RDtNZZgUuQ@mail.gmail.com>
References: <COL002-W92114B15545FE508BD1E5BF8B60@phx.gbl>,
	<CAAdTMa86HDfzM-KKBR7v3Ey35WL1-mN8wSbodh10RDtNZZgUuQ@mail.gmail.com>
Message-ID: <COL002-W1706C897CC41BCC93F07A2F8B60@phx.gbl>

My provider allow 100 channels at a time.

Date: Wed, 15 Aug 2012 06:31:59 -0500
From: asaad2 at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio with two PSTN leg

How many channels is your provider allowing you? I remember I had the same problem with asterisk a long time ago and it turned out my provider only allowed 2 channels at a time

On Aug 15, 2012 2:21 AM, "ZhouJianhua" <jh.zhou at outlook.com> wrote:




Hi,
I tried to bridge two PSTN call like this:
  originate sofia/gateway/mygateway/num1 &bridge(sofia/gateway/mygateway/num2)

Two phone ringed but no audio. There is audio if one leg is SIP user, like this:

  originate sofia/internal/1000 &bridge(sofia/gateway/mygateway/num2)

or
   originate sofia/gateway/mygateway/num1 &bridge(sofia/internal/1000)
 		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.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




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 itispip-qq at hotmail.com  Wed Aug 15 18:30:22 2012
From: itispip-qq at hotmail.com (Pip Live)
Date: Wed, 15 Aug 2012 22:30:22 +0800
Subject: [Freeswitch-users] How to config FS to listen to early media at 180?
Message-ID: <BLU0-SMTP89FB512A88818BB97CCC678CB60@phx.gbl>

Hi, my FS for outbound PSTN gateway is configed as:

(in sofia.conf xml)
<param name="inbound-bypass-media" value="false"/>
<param name="inbound-late-negotiation" value="true"/>
  .....
(in particular gateway.conf xml)
 <action application="set" data="ignore_early_media=false"/>
 <action application="set" data="bridge_early_media=true"/>
 <action application="bridge" data="sofia/gateway/PSTN1/$1"/>

But outbound call can not hear the early media played by destination number.

I checked Wiki which says FS will connect media when destination send 183,
and 180 means end-points should generate the faked ring tone; but in part
of Europ (Germany) & Asia countries, PSTN operators send early media with
180. Checked logger, it did showed a response of 180.

So is there anyway to config FS to connect media at 180 response from
destination?

Rgds, Pip
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From avi at avimarcus.net  Wed Aug 15 23:17:35 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 15 Aug 2012 22:17:35 +0300
Subject: [Freeswitch-users] How to config FS to listen to early media at
	180?
In-Reply-To: <BLU0-SMTP89FB512A88818BB97CCC678CB60@phx.gbl>
References: <BLU0-SMTP89FB512A88818BB97CCC678CB60@phx.gbl>
Message-ID: <CALNZuw=fAp851t8gT_tybbbD33ufXnmV+fFQZN+YZ75Mc325Lw@mail.gmail.com>

An fs_cli log and a pcap of the call *might* help.
-Avi
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From anton.jugatsu at gmail.com  Wed Aug 15 23:20:12 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Wed, 15 Aug 2012 23:20:12 +0400
Subject: [Freeswitch-users] IP authentification
In-Reply-To: <502B867A.5060608@pripojtese.net>
References: <502B867A.5060608@pripojtese.net>
Message-ID: <CANBhyPDRMthxhb-YiVTx-DY9K0Hn7oSa+HAYO=Oq7ucu4sMKSA@mail.gmail.com>

You should try using ACL, just search wiki.
15.08.2012 23:09 ???????????? "Jakub Tencl" <info at pripojtese.net> ???????:

> Hi,
>
> i'm in the middle of the testing our fresh installation and now i can't
> find out howto set IP authentification, when freeswitch are registered
> on the GW, it works fine but without registration, just based on IP i
> have no idea how the freeswitch is doing that if so. Can you help me
> please? I am using the vBilling system.
>
> Thanks
>
> Jakub
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 paul at cupis.co.uk  Thu Aug 16 00:06:31 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Wed, 15 Aug 2012 21:06:31 +0100
Subject: [Freeswitch-users] How to config FS to listen to early media at
 180?
In-Reply-To: <BLU0-SMTP89FB512A88818BB97CCC678CB60@phx.gbl>
References: <BLU0-SMTP89FB512A88818BB97CCC678CB60@phx.gbl>
Message-ID: <502C0147.6060409@cupis.co.uk>

On 15/08/12 15:30, Pip Live wrote:
> I checked Wiki which says FS will connect media when destination send
> 183, and 180 means end-points should generate the faked ring tone; but
> in part of Europ (Germany) & Asia countries, PSTN operators send early
> media with 180. Checked logger, it did showed a response of 180.

I would suggest logging a fault with your providers, as early media on 
180 sounds like a bug in their implementation to me.

Regards,


From krice at freeswitch.org  Thu Aug 16 03:17:56 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 15 Aug 2012 18:17:56 -0500
Subject: [Freeswitch-users] Profiling FreeSWITCH on Centos
Message-ID: <CC519854.3C0A2%krice@freeswitch.org>

Hey Guys,

Help me remember. I was speaking with one of you at ClueCon about profiling
FreeSWITCH on Centos5 and Centos6 so we can compare whats going on. I can
not remember who that was! If it was you, please contact me on IRC (I?m SwK
on #freeswitch on freenode), or Email me off list

Thanks
Ken
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From arnuld at phonologies.com  Thu Aug 16 10:22:42 2012
From: arnuld at phonologies.com (Arnuld Uttre (Phonologies))
Date: Thu, 16 Aug 2012 11:52:42 +0530
Subject: [Freeswitch-users] Regarding outgoing call
Message-ID: <63941d3bc07dadf24bf74e7a9549d4b3.squirrel@webmail1.web.com>

Currently I am making an outgoing call using "api originate" over telnet
login. I am subscribing to:

 1)  CHANNEL_CREATE  to know when a call is started.
 2)  CHANNEL_DESTROY to know that call is over and all the resources are
freed.

is this the right method ?

NOTE: someone told me I should use CHANNEL_ORIGINATE to know that a call
is started and CHANNEL_HANGUP_COMPLETE to know when call is over and all
resources are destroyed. Wiki does not have much information on this. That
is why I posted this question.


-- 
Arnuld Uttre
Systems Software Engineer

arnuld at Phonologies.COM
http://www.phonologies.com

Phonologies (India) Private Limited
West Wing, Marri Deep, M. C. H. No. 12-5-4,
Lallaguda, Secunderabad 500017, INDIA.
Ph:+91-40-2701 8993 / 36
Fax:+91-40-2701 8992




From vaad.fabi at gmail.com  Thu Aug 16 13:15:42 2012
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Thu, 16 Aug 2012 12:15:42 +0300
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <4DB00624.2010009@tagnet.ru>
References: <4DB00624.2010009@tagnet.ru>
Message-ID: <502CBA3E.7030107@gmail.com>

  Hi all,

Please suggest mobile voip dialer vendor for 
iPhone,Android,Symbian,Blackberry and WinMobile platforms.
don't suggest google plz

-- 
Best Regards,
Vadim F.



From monemran at gmail.com  Thu Aug 16 13:22:04 2012
From: monemran at gmail.com (Mohammad Emran)
Date: Thu, 16 Aug 2012 15:22:04 +0600
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <502CBA3E.7030107@gmail.com>
References: <4DB00624.2010009@tagnet.ru> <502CBA3E.7030107@gmail.com>
Message-ID: <86E0D1C3-8B78-4A71-9BA4-1C901B40FCDD@gmail.com>

Try www.sipmobiledialer.com


Sent from my iPad

On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com" <vaad.fabi at gmail.com> wrote:

>  Hi all,
> 
> Please suggest mobile voip dialer vendor for 
> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
> don't suggest google plz
> 
> -- 
> Best Regards,
> Vadim F.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 vaad.fabi at gmail.com  Thu Aug 16 13:42:03 2012
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Thu, 16 Aug 2012 12:42:03 +0300
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <86E0D1C3-8B78-4A71-9BA4-1C901B40FCDD@gmail.com>
References: <4DB00624.2010009@tagnet.ru> <502CBA3E.7030107@gmail.com>
	<86E0D1C3-8B78-4A71-9BA4-1C901B40FCDD@gmail.com>
Message-ID: <502CC06B.8070202@gmail.com>

  Seems to be ok for a simple dialer, but i forgot to mention - i need 
advanced options for dialers, such as: IM (xmpp\facebook\twitter 
integration), video calls, balance state, check direction rate, 
callback, sms, top up and etc. It's smth like UC application with 
calling support as one of the many options.


On 08/16/2012 12:22 PM, Mohammad Emran wrote:
> Try www.sipmobiledialer.com
>
>
> Sent from my iPad
>
> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com"<vaad.fabi at gmail.com>  wrote:
>
>>   Hi all,
>>
>> Please suggest mobile voip dialer vendor for
>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>> don't suggest google plz
>>
>> -- 
>> Best Regards,
>> Vadim F.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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


-- 
Best Regards,
Vadim F.



From monemran at gmail.com  Thu Aug 16 13:53:43 2012
From: monemran at gmail.com (Mohammad Emran)
Date: Thu, 16 Aug 2012 15:53:43 +0600
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <502CC06B.8070202@gmail.com>
References: <4DB00624.2010009@tagnet.ru> <502CBA3E.7030107@gmail.com>
	<86E0D1C3-8B78-4A71-9BA4-1C901B40FCDD@gmail.com>
	<502CC06B.8070202@gmail.com>
Message-ID: <806E15A7-FD82-4215-B50E-9B921CB2DC7D@gmail.com>

Then try nimbuzz or fring.... If it works for u.

Sent from my iPhone

On Aug 16, 2012, at 3:42 PM, "vaad.fabi at gmail.com" <vaad.fabi at gmail.com> wrote:

>  Seems to be ok for a simple dialer, but i forgot to mention - i need 
> advanced options for dialers, such as: IM (xmpp\facebook\twitter 
> integration), video calls, balance state, check direction rate, 
> callback, sms, top up and etc. It's smth like UC application with 
> calling support as one of the many options.
> 
> 
> On 08/16/2012 12:22 PM, Mohammad Emran wrote:
>> Try www.sipmobiledialer.com
>> 
>> 
>> Sent from my iPad
>> 
>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com"<vaad.fabi at gmail.com>  wrote:
>> 
>>>  Hi all,
>>> 
>>> Please suggest mobile voip dialer vendor for
>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>>> don't suggest google plz
>>> 
>>> -- 
>>> Best Regards,
>>> Vadim F.
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> 
> -- 
> Best Regards,
> Vadim F.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 devel at omninet.eu  Thu Aug 16 14:10:58 2012
From: devel at omninet.eu (Anestis Mavro)
Date: Thu, 16 Aug 2012 13:10:58 +0300
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <502CC06B.8070202@gmail.com>
References: <4DB00624.2010009@tagnet.ru>
	<502CBA3E.7030107@gmail.com><86E0D1C3-8B78-4A71-9BA4-1C901B40FCDD@gmail.com>
	<502CC06B.8070202@gmail.com>
Message-ID: <EF69A1CC639B43C7BB76BC5FB79A258E@omni1.local>


It looks like you need a lot on customization.
Ask the companies listed on the interop page.

I am working a lot with Media5-Fone without problems:
http://wiki.freeswitch.org/wiki/Interop_List#Media5Fone_for_iPhone

Regards,


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
vaad.fabi at gmail.com
Sent: Thursday, August 16, 2012 12:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mobile dialers

  Seems to be ok for a simple dialer, but i forgot to mention - i need 
advanced options for dialers, such as: IM (xmpp\facebook\twitter 
integration), video calls, balance state, check direction rate, 
callback, sms, top up and etc. It's smth like UC application with 
calling support as one of the many options.


On 08/16/2012 12:22 PM, Mohammad Emran wrote:
> Try www.sipmobiledialer.com
>
>
> Sent from my iPad
>
> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com"<vaad.fabi at gmail.com>
wrote:
>
>>   Hi all,
>>
>> Please suggest mobile voip dialer vendor for
>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>> don't suggest google plz
>>
>> -- 
>> Best Regards,
>> Vadim F.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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


-- 
Best Regards,
Vadim F.


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

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

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 5054 (20100423) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
 
 

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 5054 (20100423) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
 



From ben at langfeld.co.uk  Thu Aug 16 14:39:49 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Thu, 16 Aug 2012 11:39:49 +0100
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <502CBA3E.7030107@gmail.com>
References: <4DB00624.2010009@tagnet.ru>
	<502CBA3E.7030107@gmail.com>
Message-ID: <CAAyX+KEb0tsof52qOF15V3pOxT7pB230DkiDyDfZ4Skuiuvf7g@mail.gmail.com>

The term you're looking for is 'softphone'. A 'dialer' is a very different
beast, and trying to get one to perform well on smartphone hardware would
be....interesting.

Regards,
Ben Langfeld


On 16 August 2012 10:15, vaad.fabi at gmail.com <vaad.fabi at gmail.com> wrote:

>   Hi all,
>
> Please suggest mobile voip dialer vendor for
> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
> don't suggest google plz
>
> --
> Best Regards,
> Vadim F.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
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From vaad.fabi at gmail.com  Thu Aug 16 15:11:56 2012
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Thu, 16 Aug 2012 14:11:56 +0300
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <CAAyX+KEb0tsof52qOF15V3pOxT7pB230DkiDyDfZ4Skuiuvf7g@mail.gmail.com>
References: <4DB00624.2010009@tagnet.ru>	<502CBA3E.7030107@gmail.com>
	<CAAyX+KEb0tsof52qOF15V3pOxT7pB230DkiDyDfZ4Skuiuvf7g@mail.gmail.com>
Message-ID: <502CD57C.1090701@gmail.com>

  ok, let's it be "UC application"  :)


On 08/16/2012 01:39 PM, Ben Langfeld wrote:
> The term you're looking for is 'softphone'. A 'dialer' is a very 
> different beast, and trying to get one to perform well on smartphone 
> hardware would be....interesting.
>
> Regards,
> Ben Langfeld
>
>
> On 16 August 2012 10:15, vaad.fabi at gmail.com 
> <mailto:vaad.fabi at gmail.com> <vaad.fabi at gmail.com 
> <mailto:vaad.fabi at gmail.com>> wrote:
>
>       Hi all,
>
>     Please suggest mobile voip dialer vendor for
>     iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>     don't suggest google plz
>
>     --
>     Best Regards,
>     Vadim F.
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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


-- 
Best Regards,
Vadim F.

-------------- next part --------------
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From vaad.fabi at gmail.com  Thu Aug 16 15:12:45 2012
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Thu, 16 Aug 2012 14:12:45 +0300
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <EF69A1CC639B43C7BB76BC5FB79A258E@omni1.local>
References: <4DB00624.2010009@tagnet.ru>	<502CBA3E.7030107@gmail.com><86E0D1C3-8B78-4A71-9BA4-1C901B40FCDD@gmail.com>	<502CC06B.8070202@gmail.com>
	<EF69A1CC639B43C7BB76BC5FB79A258E@omni1.local>
Message-ID: <502CD5AD.7070302@gmail.com>

  thx. i'll try this.

On 08/16/2012 01:10 PM, Anestis Mavro wrote:
> It looks like you need a lot on customization.
> Ask the companies listed on the interop page.
>
> I am working a lot with Media5-Fone without problems:
> http://wiki.freeswitch.org/wiki/Interop_List#Media5Fone_for_iPhone
>
> Regards,
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> vaad.fabi at gmail.com
> Sent: Thursday, August 16, 2012 12:42 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mobile dialers
>
>    Seems to be ok for a simple dialer, but i forgot to mention - i need
> advanced options for dialers, such as: IM (xmpp\facebook\twitter
> integration), video calls, balance state, check direction rate,
> callback, sms, top up and etc. It's smth like UC application with
> calling support as one of the many options.
>
>
> On 08/16/2012 12:22 PM, Mohammad Emran wrote:
>> Try www.sipmobiledialer.com
>>
>>
>> Sent from my iPad
>>
>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com"<vaad.fabi at gmail.com>
> wrote:
>>>    Hi all,
>>>
>>> Please suggest mobile voip dialer vendor for
>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>>> don't suggest google plz
>>>
>>> -- 
>>> Best Regards,
>>> Vadim F.
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>


-- 
Best Regards,
Vadim F.



From vaad.fabi at gmail.com  Thu Aug 16 15:16:45 2012
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Thu, 16 Aug 2012 14:16:45 +0300
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <806E15A7-FD82-4215-B50E-9B921CB2DC7D@gmail.com>
References: <4DB00624.2010009@tagnet.ru>
	<502CBA3E.7030107@gmail.com>	<86E0D1C3-8B78-4A71-9BA4-1C901B40FCDD@gmail.com>	<502CC06B.8070202@gmail.com>
	<806E15A7-FD82-4215-B50E-9B921CB2DC7D@gmail.com>
Message-ID: <502CD69D.1060908@gmail.com>

  Does they provide these software development, customisation and 
support ? i think no.  I need vendor\development team who can provide 
development\customisation\support.
Anyway thx.


On 08/16/2012 12:53 PM, Mohammad Emran wrote:
> Then try nimbuzz or fring.... If it works for u.
>
> Sent from my iPhone
>
> On Aug 16, 2012, at 3:42 PM, "vaad.fabi at gmail.com"<vaad.fabi at gmail.com>  wrote:
>
>>   Seems to be ok for a simple dialer, but i forgot to mention - i need
>> advanced options for dialers, such as: IM (xmpp\facebook\twitter
>> integration), video calls, balance state, check direction rate,
>> callback, sms, top up and etc. It's smth like UC application with
>> calling support as one of the many options.
>>
>>
>> On 08/16/2012 12:22 PM, Mohammad Emran wrote:
>>> Try www.sipmobiledialer.com
>>>
>>>
>>> Sent from my iPad
>>>
>>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com"<vaad.fabi at gmail.com>   wrote:
>>>
>>>>   Hi all,
>>>>
>>>> Please suggest mobile voip dialer vendor for
>>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>>>> don't suggest google plz
>>>>
>>>> -- 
>>>> Best Regards,
>>>> Vadim F.
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>
>> -- 
>> Best Regards,
>> Vadim F.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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


-- 
Best Regards,
Vadim F.



From m.prevail at gmail.com  Thu Aug 16 16:29:23 2012
From: m.prevail at gmail.com (Prevail Magid)
Date: Thu, 16 Aug 2012 15:29:23 +0300
Subject: [Freeswitch-users] 3 sessions instead of one. with use loopback.
Message-ID: <CANJXrA-ZaKcSvmpvJJOMgouovD-08NmGe8zDxO2R_JY5d7GvLw@mail.gmail.com>

Hi,

I have ESL client that send comand to FS for invite person to conference
like : "conference 777 dial loopback/ADD_CONF_1007"

and dialplan for catch this :

       <condition field="destination_number"
expression="^ADD_CONF_(\d{4})$" break="on-true">
                        <action application="set"
data="hangup_after_bridge=true" />
<action application="bridge" data="user/$1" />
</condition>

and I have three channels :  loopback/ADD_CONF-a and loopback/ADD_CONF-b -
(that`s created loopback?) and sofia/internal/sip...

How can I close loopback channels, thats do not need as understand after
bridge?
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From asaad2 at gmail.com  Thu Aug 16 16:44:54 2012
From: asaad2 at gmail.com (BookBag)
Date: Thu, 16 Aug 2012 07:44:54 -0500
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <502CD69D.1060908@gmail.com>
References: <4DB00624.2010009@tagnet.ru> <502CBA3E.7030107@gmail.com>
	<86E0D1C3-8B78-4A71-9BA4-1C901B40FCDD@gmail.com>
	<502CC06B.8070202@gmail.com>
	<806E15A7-FD82-4215-B50E-9B921CB2DC7D@gmail.com>
	<502CD69D.1060908@gmail.com>
Message-ID: <CAAdTMa9=+4bQtJ75NAR+uDckFW8Q531iMSwWo17nYjbx9co5CQ@mail.gmail.com>

I'm surprised no one said Bria by counterpath. Which allows for xmpp and
presence. The softphone allows you to send sms also if you buy the full
version. Another one that I tested is Csipsimple, this one allows sms for
free. But my absolute favorite is Zoiper, unfortunately no sms capability
but wasnt buggy except for one bug I encountered with sending the correct
dtmf style. Thank god their new update took care of that. Phone now seems
very solid and now includes a QR reader.

On Thu, Aug 16, 2012 at 6:16 AM, vaad.fabi at gmail.com <vaad.fabi at gmail.com>wrote:

>   Does they provide these software development, customisation and
> support ? i think no.  I need vendor\development team who can provide
> development\customisation\support.
> Anyway thx.
>
>
> On 08/16/2012 12:53 PM, Mohammad Emran wrote:
> > Then try nimbuzz or fring.... If it works for u.
> >
> > Sent from my iPhone
> >
> > On Aug 16, 2012, at 3:42 PM, "vaad.fabi at gmail.com"<vaad.fabi at gmail.com>
>  wrote:
> >
> >>   Seems to be ok for a simple dialer, but i forgot to mention - i need
> >> advanced options for dialers, such as: IM (xmpp\facebook\twitter
> >> integration), video calls, balance state, check direction rate,
> >> callback, sms, top up and etc. It's smth like UC application with
> >> calling support as one of the many options.
> >>
> >>
> >> On 08/16/2012 12:22 PM, Mohammad Emran wrote:
> >>> Try www.sipmobiledialer.com
> >>>
> >>>
> >>> Sent from my iPad
> >>>
> >>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com"<vaad.fabi at gmail.com>
>   wrote:
> >>>
> >>>>   Hi all,
> >>>>
> >>>> Please suggest mobile voip dialer vendor for
> >>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
> >>>> don't suggest google plz
> >>>>
> >>>> --
> >>>> Best Regards,
> >>>> Vadim F.
> >>>>
> >>>>
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.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
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.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
> >>
> >> --
> >> Best Regards,
> >> Vadim F.
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
>
> --
> Best Regards,
> Vadim F.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 asaad2 at gmail.com  Thu Aug 16 16:48:18 2012
From: asaad2 at gmail.com (BookBag)
Date: Thu, 16 Aug 2012 07:48:18 -0500
Subject: [Freeswitch-users] Limitation
In-Reply-To: <502B8B67.3090306@pripojtese.net>
References: <502B8B67.3090306@pripojtese.net>
Message-ID: <CAAdTMa-CEoYQm_fAkOiSAsRG9Dvio8vx8ABUvOZTTE2LyAN8xQ@mail.gmail.com>

I hope the FusionPbx team is reading this. All these features are available
in FusionPbx but they havent released a version which allows billing. They
said they would in the next release.

On Wed, Aug 15, 2012 at 6:43 AM, Jakub Tencl <info at pripojtese.net> wrote:

> Hello,
>
> i'm just wondering what is the limitation in freeswitch and vBilling
> system, i've noticed in the log that i can make 250 concurrent calls and
> 50k calls per day and if i am using authentification via IP, is there
> another limitation? Here is log from my testing call:
>
> ------------------------------------------------------------------------
> send 321 bytes to udp/[x.x.x.x]:5060 at 11:13:48.500060:
> ------------------------------------------------------------------------
>     ACK sip:phone number at x.x.x.x SIP/2.0
>     Via: SIP/2.0/UDP x.x.x.x;rport;branch=z9hG4bKXv02SBvK3U8SS
>     Max-Forwards: 69
>     From: "281545" <sip:281545 at x.x.x.x>;tag=DDNBjUggvFm6B
>     To: <sip:phone number at x.x.x.x>
>     Call-ID: 20a033f8-616d-1230-0e88-00188b8cb036
>     CSeq: 32178550 ACK
>     Content-Length: 0
>
> ------------------------------------------------------------------------
> 2012-08-15 12:13:48.496981 [NOTICE] sofia.c:6847 Hangup
> sofia/Billing205/phone number [CS_CONSUME_MEDIA] [CALL_REJECTED]
> 2012-08-15 12:13:48.496981 [NOTICE] switch_core_session.c:1447 Session
> 146 (sofia/Billing205/phone number) Ended
> 2012-08-15 12:13:48.496981 [INFO] mod_dptools.c:3027 Originate Failed.
> Cause: CALL_REJECTED
> 2012-08-15 12:13:48.496981 [NOTICE] switch_core_session.c:1449 Close
> Channel sofia/Billing205/phone number [CS_DESTROY]
> 2012-08-15 12:13:48.496981 [NOTICE] mod_dptools.c:3147 Hangup
> sofia/Billing205/281545 at x.x.x.x:5060 [CS_EXECUTE] [CALL_REJECTED]
> 2012-08-15 12:13:48.496981 [INFO] mod_hash.c:304 Usage for
> x.x.x.x_192.168.2.96 is now 0
> 2012-08-15 12:13:48.496981 [INFO] mod_hash.c:304 Usage for
> MAX_CALLS_PER_DAY_IN_FREE_MODE_call_per_day is now 0
> 2012-08-15 12:13:48.496981 [INFO] mod_hash.c:304 Usage for
> CONCURRENT_CALLS_IN_FREE_MODE_calls_max is now 0
> send 752 bytes to udp/[192.168.2.96]:54205 at 11:13:48.502357:
> ------------------------------------------------------------------------
>     SIP/2.0 401 Unauthorized
>     Via: SIP/2.0/UDP
> 192.168.5.110:54205
> ;branch=z9hG4bK-d8754z-4b7558087e1e491d-1---d8754z-;rport=54205;received=192.168.2.96
>     Max-Forwards: 70
>     From: "281545"<sip:281545 at x.x.x.x:5060>;tag=db146842
>     To: <sip:phone number at x.x.x.x:5060>;tag=c4Ujg0ZcZ6XKg
>     Call-ID: ZDM4YTU2ZmU3ZjNhNTc2MzAyZGZiYmNmMmVjYTRiNzE.
>     CSeq: 2 INVITE
>     User-Agent: vBilling
>     Accept: application/sdp
>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, NOTIFY
>     Supported: precondition, path, replaces
>     Allow-Events: talk, hold, conference, refer
>     Reason: Q.850;cause=21;text="CALL_REJECTED"
>     Content-Length: 0
>     Remote-Party-ID: "phone number" <sip:phone
> number at x.x.x.x>;party=calling;privacy=off;screen=no
>
> ------------------------------------------------------------------------
> 2012-08-15 12:13:48.496981 [NOTICE] switch_core_session.c:1447 Session
> 145 (sofia/Billing205/281545 at x.x.x.x:5060) Ended
> 2012-08-15 12:13:48.496981 [NOTICE] switch_core_session.c:1449 Close
> Channel sofia/Billing205/281545 at x.x.x.x:5060 [CS_DESTROY]
>
> Thanks
>
> Jakub
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 brett at launch3.net  Thu Aug 16 18:09:33 2012
From: brett at launch3.net (Brett Wilson)
Date: Thu, 16 Aug 2012 10:09:33 -0400
Subject: [Freeswitch-users] Dropping calls after 40 secs
Message-ID: <cbb3a1a7-d000-4a3c-81d2-cfc7c600bce5.maildroid@localhost>

hello,
I am having issues with calls being dropped after about 40 secs. I just compiled and installed the latest fs. Here is log output: pastebin.com/abJJfLUi 

******************************************* 
Brett Wilson 
IT Department
Launch 3 Ventures, LLC 
134 Myer Street 
Hackensack, NJ 07601 
Phone: 877.878.9134 
Fax: 646.536.3866 
Cell: 973.650.7189
Email: brett at launch3.net 
www.Launch3.net 
www.Launch3telecom.com *******************************************

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From anton.jugatsu at gmail.com  Thu Aug 16 18:19:58 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Thu, 16 Aug 2012 18:19:58 +0400
Subject: [Freeswitch-users] Dropping calls after 40 secs
In-Reply-To: <cbb3a1a7-d000-4a3c-81d2-cfc7c600bce5.maildroid@localhost>
References: <cbb3a1a7-d000-4a3c-81d2-cfc7c600bce5.maildroid@localhost>
Message-ID: <CANBhyPANkTy5YEe2uCbwFCGSCqFELjKt9H4_D8-rs3yB-LAG0g@mail.gmail.com>

Must be NAT, but we can not tell you exactly cause you didn't provide any
logs.
16.08.2012 18:11 ???????????? "Brett Wilson" <brett at launch3.net> ???????:

> hello,
> I am having issues with calls being dropped after about 40 secs. I just
> compiled and installed the latest fs. Here is log output:
> pastebin.com/abJJfLUi
>
> *******************************************
> *Brett Wilson *
> IT Department
> Launch 3 Ventures, LLC
> 134 Myer Street
> Hackensack, NJ 07601
> *Phone:* 877.878.9134 <8778789134>
> *Fax:* 646.536.3866 <6465363866>
> *Cell:* 973.650.7189 <9736507189>
> *Email:* brett at launch3.net
> www.Launch3.net
> www.Launch3telecom.com *******************************************
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ntomer at newgen.co.in  Thu Aug 16 15:40:33 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Thu, 16 Aug 2012 17:10:33 +0530
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
Message-ID: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>

Hi,

 

I am working on a Contact Center solution. It will support mail, chat and
call queries.

The requirements are:

1. An end-customer calls, the call is handled by FreeSWITCH Auto Attendant.
2. Customer is presented with a menu and makes selection. His call is put on
hold and an entry is made in my system's database for incoming queries.
3. These queries are shown to agents handling calls.
4. And agent clicks on a query, he is shown an extension where call is
parked. He dials that and is connected to the customer.

5. He talks to the customer and resolve his queries.

Please guide me on how can do it. The application will be written in Java.

I am an experienced programmers in Java/J2EE; but doesn't have much
knowledge of VoIP/FreeSWITCH. I've configured FreeSWITCH on Ubuntu 12.04;
and have installed X-Lite Softphones on two Windows machines. I configured
these phones to work with FreeSWITCH and they are working fine.

 

I think that I'd have to use Valet Parking to park customer's call on an
extension, then pass that extension to agent who will dial it and will be
connected to the customer.

 

Please tell me whether this approach will work? And how should I go about
it.

 

All help would be much appreciated.

 

Regards

 

Nitin

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From lists at kavun.ch  Thu Aug 16 16:06:15 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 16 Aug 2012 08:06:15 -0400
Subject: [Freeswitch-users] Emulate nat=yes
Message-ID: <A31A0D28-6E24-40F8-8886-5B8D578A2A66@kavun.ch>

Hello,

I am having trouble connecting my Polycom phone to FS. It works fine with Asterisk and I am trying to emulate the nat=yes behavior over on FreeSWITCH.

Both the Asterisk and FS servers are public. I am using the default FreeSWITCH sample configuration and attempting to register on the internal profile.

I have tried using NDLB-connectile-dysfunction with no luck. Any suggestions of what else I could try?

Directions to CLI commands to debug SIP would be much appreciated as well. 

Thanks a bunch for your help.


--
Emrah

?In theory, theory and practice are the same. In practice, they are not.? Albert Einstein



From obiebrown at gmail.com  Thu Aug 16 17:53:30 2012
From: obiebrown at gmail.com (Obie Brown)
Date: Thu, 16 Aug 2012 23:23:30 +0930
Subject: [Freeswitch-users] FreeSWITCH Auto Dialer - Open source
Message-ID: <CA+aeUL_D-1sgqxvxshU9oqELh66Ck-_kTxesMCCjnv3DgDu-yg@mail.gmail.com>

Hi,

I really want to work with an Open Source dialer and FreeSWITCH but the
only thing I can find for this is Newfies Dialer which does not have
features of a Auto Dialer as I need. Also its use for Voice Broadcasting is
limited.

My question is. What other Auto Dialers that run on FreeSWITCH are there.
If any.


Thanks,
Obie Brown
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From lynn.nielson at greenseedtechnologies.com  Thu Aug 16 18:54:53 2012
From: lynn.nielson at greenseedtechnologies.com (Lynn Nielson)
Date: Thu, 16 Aug 2012 08:54:53 -0600
Subject: [Freeswitch-users] uuid_broadcast and fifo
Message-ID: <502D09BD.2020208@greenseedtechnologies.com>

Is it possible to do a uuid_broadcast into a fifo parked call?  At the 
command line it tells me the Message is Sent but I do not hear it on the 
call.

The equivalent commands I'm using from fs_cli are as follows:

    originate sofia/gateway/flowroute/<external 10digit number>
    &fifo('myque out wait')

    uuid_broadcast <uuid from previous command> <path_to_audio_file>


I know that I can have an audio file playing within the fifo arguments 
but was desiring more interactive control that uuid_broadcast looked 
like it could provide.


Thanks,

Lynn

-- 
--
Lynn Nielson
Green Seed Technologies / The RedX

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From krice at freeswitch.org  Thu Aug 16 19:04:50 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 16 Aug 2012 10:04:50 -0500
Subject: [Freeswitch-users] FreeSWITCH Auto Dialer - Open source
In-Reply-To: <CA+aeUL_D-1sgqxvxshU9oqELh66Ck-_kTxesMCCjnv3DgDu-yg@mail.gmail.com>
Message-ID: <CC527642.3C0C8%krice@freeswitch.org>

Why not get with Areski, and see where you can help him instead of
completely re-inventing the wheel... Other wise, I don?t think there are
other real auto-dialers out there


On 8/16/12 8:53 AM, "Obie Brown" <obiebrown at gmail.com> wrote:

> Hi,
> 
> I really want to work with an Open Source dialer and FreeSWITCH but the only
> thing I can find for this is Newfies Dialer which does not have features of a
> Auto Dialer as I need. Also its use for Voice Broadcasting is limited.
> 
> My question is. What other Auto Dialers that run on FreeSWITCH are there. If
> any.
> 
> 
> Thanks,
> Obie Brown
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 abaci64 at gmail.com  Thu Aug 16 19:07:24 2012
From: abaci64 at gmail.com (Abaci)
Date: Thu, 16 Aug 2012 11:07:24 -0400
Subject: [Freeswitch-users] FreeSWITCH Auto Dialer - Open source
In-Reply-To: <CA+aeUL_D-1sgqxvxshU9oqELh66Ck-_kTxesMCCjnv3DgDu-yg@mail.gmail.com>
References: <CA+aeUL_D-1sgqxvxshU9oqELh66Ck-_kTxesMCCjnv3DgDu-yg@mail.gmail.com>
Message-ID: <502D0CAC.8030905@gmail.com>

http://www.ictdialer.org/

On 8/16/2012 9:53 AM, Obie Brown wrote:
> Hi,
>
> I really want to work with an Open Source dialer and FreeSWITCH but 
> the only thing I can find for this is Newfies Dialer which does not 
> have features of a Auto Dialer as I need. Also its use for Voice 
> Broadcasting is limited.
>
> My question is. What other Auto Dialers that run on FreeSWITCH are 
> there. If any.
>
>
> Thanks,
> Obie Brown
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Thu Aug 16 19:11:18 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Aug 2012 08:11:18 -0700
Subject: [Freeswitch-users] Profiling FreeSWITCH on Centos
In-Reply-To: <CC519854.3C0A2%krice@freeswitch.org>
References: <CC519854.3C0A2%krice@freeswitch.org>
Message-ID: <CAKzWOxURgEuVWtQSSm7NxMBS7Z4c-OmcsrPTGD9Lb6eEj0154g@mail.gmail.com>

Was it Marcel B. over at Audio Now?
-MC

On Wed, Aug 15, 2012 at 4:17 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Hey Guys,
>
> Help me remember. I was speaking with one of you at ClueCon about
> profiling FreeSWITCH on Centos5 and Centos6 so we can compare whats going
> on. I can not remember who that was! If it was you, please contact me on
> IRC (I?m SwK on #freeswitch on freenode), or Email me off list
>
> Thanks
> Ken
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From brett at launch3.net  Thu Aug 16 19:31:44 2012
From: brett at launch3.net (Brett Wilson)
Date: Thu, 16 Aug 2012 11:31:44 -0400
Subject: [Freeswitch-users] Dropping calls after 40 secs
In-Reply-To: <CANBhyPANkTy5YEe2uCbwFCGSCqFELjKt9H4_D8-rs3yB-LAG0g@mail.gmail.com>
References: <cbb3a1a7-d000-4a3c-81d2-cfc7c600bce5.maildroid@localhost>
	<CANBhyPANkTy5YEe2uCbwFCGSCqFELjKt9H4_D8-rs3yB-LAG0g@mail.gmail.com>
Message-ID: <005801cd7bc4$3e394700$baabd500$@launch3.net>

What do you need logs of? I posted that pastebin link for the log output on
max verbosity. I do have auto-nat enabled.

 

******************************************* 
Brett Wilson
IT Department
Launch 3 Ventures, LLC 
134 Myer Street 
Hackensack, NJ 07601 
Phone: 877.878.9134
Fax: 646.536.3866 
Email: Brett.Wilson at launch3.net
AOL IM: Brett.Wilson at launch3.net
 <http://www.launch3.net/> www.Launch3.net 
www.Launch3telecom.com <http://www.launch3telecom.com/>  
******************************************* 
 <http://launch3telecom.blogspot.com/> Description: Description:
Description: Blogger-logo
<http://www.facebook.com/pages/Launch-3-Telecom/165610820178720>
Description: Description: Description: FaceBook-Logo
<https://twitter.com/launch3telecom> Description: Description: Description:
Twitter-Logo <https://plus.google.com/102713212642017290959> Description:
Description: Description: GPlus-Logo

 

From: Anton Kvashenkin [mailto:anton.jugatsu at gmail.com] 
Sent: Thursday, August 16, 2012 10:20 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping calls after 40 secs

 

Must be NAT, but we can not tell you exactly cause you didn't provide any
logs.

16.08.2012 18:11 ???????????? "Brett Wilson" <brett at launch3.net> ???????:

hello,
I am having issues with calls being dropped after about 40 secs. I just
compiled and installed the latest fs. Here is log output:
pastebin.com/abJJfLUi 

******************************************* 
Brett Wilson 
IT Department
Launch 3 Ventures, LLC 
134 Myer Street 
Hackensack, NJ 07601 
Phone: 877.878.9134 <tel:8778789134>  
Fax: 646.536.3866 <tel:6465363866>  
Cell: 973.650.7189 <tel:9736507189> 
Email: brett at launch3.net 
www.Launch3.net 
www.Launch3telecom.com *******************************************


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 krice at freeswitch.org  Thu Aug 16 19:37:29 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 16 Aug 2012 10:37:29 -0500
Subject: [Freeswitch-users] Profiling FreeSWITCH on Centos
In-Reply-To: <CAKzWOxURgEuVWtQSSm7NxMBS7Z4c-OmcsrPTGD9Lb6eEj0154g@mail.gmail.com>
Message-ID: <CC527DE9.3C0DA%krice@freeswitch.org>

It Might have been... Collins can you put me in touch with him offlist?


On 8/16/12 10:11 AM, "Michael Collins" <msc at freeswitch.org> wrote:

> Was it Marcel B. over at Audio Now?
> -MC
> 
> On Wed, Aug 15, 2012 at 4:17 PM, Ken Rice <krice at freeswitch.org> wrote:
>> Hey Guys,
>> 
>> Help me remember. I was speaking with one of you at ClueCon about profiling
>> FreeSWITCH on Centos5 and Centos6 so we can compare whats going on. I can not
>> remember who that was! If it was you, please contact me on IRC (I?m SwK on
>> #freeswitch on freenode), or Email me off list
>> 
>> Thanks
>> Ken 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 bdfoster at endigotech.com  Thu Aug 16 20:52:09 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 16 Aug 2012 12:52:09 -0400
Subject: [Freeswitch-users] Dropping calls after 40 secs
In-Reply-To: <005801cd7bc4$3e394700$baabd500$@launch3.net>
References: <cbb3a1a7-d000-4a3c-81d2-cfc7c600bce5.maildroid@localhost>
	<CANBhyPANkTy5YEe2uCbwFCGSCqFELjKt9H4_D8-rs3yB-LAG0g@mail.gmail.com>
	<005801cd7bc4$3e394700$baabd500$@launch3.net>
Message-ID: <CAKV-jFw5mrzxag-QeuCBL0=NRbH7FonPajshNuWEth6=waGNRg@mail.gmail.com>

Need the siptrace of all legs involved, combined with the debug that you
posted. We need to be able to look at the sip messages involved to find
clues.

sofia profile internal siptrace on
sofia profile external siptrace on

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 16, 2012 11:36 AM, "Brett Wilson" <brett at launch3.net> wrote:

> What do you need logs of? I posted that pastebin link for the log output
> on max verbosity. I do have auto-nat enabled.****
>
> ** **
>
> *******************************************
> *Brett Wilson*
> *IT Department*
> *Launch 3 Ventures, LLC*
> 134 Myer Street
> Hackensack, NJ 07601
> *Phone:* 877.878.9134
> *Fax:* 646.536.3866
> *Email:* Brett.Wilson at launch3.net
> *AOL IM:* Brett.Wilson at launch3.net
> www.Launch3.net <http://www.launch3.net/>
> *www.Launch3telecom.com <http://www.launch3telecom.com/>*
> *******************************************
> [image: Description: Description: Description: Blogger-logo]<http://launch3telecom.blogspot.com/>[image:
> Description: Description: Description: FaceBook-Logo]<http://www.facebook.com/pages/Launch-3-Telecom/165610820178720>[image:
> Description: Description: Description: Twitter-Logo]<https://twitter.com/launch3telecom>[image:
> Description: Description: Description: GPlus-Logo]<https://plus.google.com/102713212642017290959>
> ****
>
> ** **
>
> *From:* Anton Kvashenkin [mailto:anton.jugatsu at gmail.com]
> *Sent:* Thursday, August 16, 2012 10:20 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Dropping calls after 40 secs****
>
> ** **
>
> Must be NAT, but we can not tell you exactly cause you didn't provide any
> logs.****
>
> 16.08.2012 18:11 ???????????? "Brett Wilson" <brett at launch3.net> ???????:*
> ***
>
> hello,
> I am having issues with calls being dropped after about 40 secs. I just
> compiled and installed the latest fs. Here is log output:
> pastebin.com/abJJfLUi ****
>
> *******************************************
> *Brett Wilson *
> IT Department
> Launch 3 Ventures, LLC
> 134 Myer Street
> Hackensack, NJ 07601
> *Phone:* 877.878.9134 <8778789134>
> *Fax:* 646.536.3866 <6465363866>
> *Cell:* 973.650.7189 <9736507189>
> *Email:* brett at launch3.net
> www.Launch3.net
> www.Launch3telecom.com ***********************************************
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sdevoy at bizfocused.com  Thu Aug 16 21:35:49 2012
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Thu, 16 Aug 2012 13:35:49 -0400
Subject: [Freeswitch-users] CALL TIMEOUT and interprise bridge
Message-ID: <00a201cd7bd5$9382dd50$ba8897f0$@bizfocused.com>

Hi,

I have noticed that "SET [call_timeout]=[25]" seems to be ignored when using
an Enterprise bridge (:_:) to two extensions.  The timeout is always 60
seconds.

 

Do I just need to set the timeout in {} on each leg?

 

Sean

 

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From freeswitch at yeah.net  Thu Aug 16 21:58:00 2012
From: freeswitch at yeah.net (freeswitch)
Date: Fri, 17 Aug 2012 01:58:00 +0800 (CST)
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <CAAdTMa9=+4bQtJ75NAR+uDckFW8Q531iMSwWo17nYjbx9co5CQ@mail.gmail.com>
References: <4DB00624.2010009@tagnet.ru> <502CBA3E.7030107@gmail.com>
	<86E0D1C3-8B78-4A71-9BA4-1C901B40FCDD@gmail.com>
	<502CC06B.8070202@gmail.com>
	<806E15A7-FD82-4215-B50E-9B921CB2DC7D@gmail.com>
	<502CD69D.1060908@gmail.com>
	<CAAdTMa9=+4bQtJ75NAR+uDckFW8Q531iMSwWo17nYjbx9co5CQ@mail.gmail.com>
Message-ID: <7357444f.60.1393095b1e5.Coremail.freeswitch@yeah.net>

IMSDROID will be good!




At 2012-08-16 20:44:54,BookBag <asaad2 at gmail.com> wrote:
I'm surprised no one said Bria by counterpath. Which allows for xmpp and presence. The softphone allows you to send sms also if you buy the full version. Another one that I tested is Csipsimple, this one allows sms for free. But my absolute favorite is Zoiper, unfortunately no sms capability but wasnt buggy except for one bug I encountered with sending the correct dtmf style. Thank god their new update took care of that. Phone now seems very solid and now includes a QR reader.


On Thu, Aug 16, 2012 at 6:16 AM, vaad.fabi at gmail.com<vaad.fabi at gmail.com> wrote:
  Does they provide these software development, customisation and
support ? i think no.  I need vendor\development team who can provide
development\customisation\support.
Anyway thx.



On 08/16/2012 12:53 PM, Mohammad Emran wrote:
> Then try nimbuzz or fring.... If it works for u.
>
> Sent from my iPhone
>
> On Aug 16, 2012, at 3:42 PM, "vaad.fabi at gmail.com"<vaad.fabi at gmail.com>  wrote:
>
>>   Seems to be ok for a simple dialer, but i forgot to mention - i need
>> advanced options for dialers, such as: IM (xmpp\facebook\twitter
>> integration), video calls, balance state, check direction rate,
>> callback, sms, top up and etc. It's smth like UC application with
>> calling support as one of the many options.
>>
>>
>> On 08/16/2012 12:22 PM, Mohammad Emran wrote:
>>> Try www.sipmobiledialer.com
>>>
>>>
>>> Sent from my iPad
>>>
>>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com"<vaad.fabi at gmail.com>   wrote:
>>>
>>>>   Hi all,
>>>>
>>>> Please suggest mobile voip dialer vendor for
>>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>>>> don't suggest google plz
>>>>
>>>> --
>>>> Best Regards,
>>>> Vadim F.
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>
>> --
>> Best Regards,
>> Vadim F.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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


--
Best Regards,
Vadim F.


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 nickolayr at gmail.com  Thu Aug 16 22:08:41 2012
From: nickolayr at gmail.com (Nikolay Rogoshchenkov)
Date: Thu, 16 Aug 2012 14:08:41 -0400
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <7357444f.60.1393095b1e5.Coremail.freeswitch@yeah.net>
References: <4DB00624.2010009@tagnet.ru> <502CBA3E.7030107@gmail.com>
	<86E0D1C3-8B78-4A71-9BA4-1C901B40FCDD@gmail.com>
	<502CC06B.8070202@gmail.com>
	<806E15A7-FD82-4215-B50E-9B921CB2DC7D@gmail.com>
	<502CD69D.1060908@gmail.com>
	<CAAdTMa9=+4bQtJ75NAR+uDckFW8Q531iMSwWo17nYjbx9co5CQ@mail.gmail.com>
	<7357444f.60.1393095b1e5.Coremail.freeswitch@yeah.net>
Message-ID: <CAAQCs31UinwDE_W9qAgyuW7Q4vEszbd3A_yAuuSN1J9-+bMnCA@mail.gmail.com>

I can also recommend *Sipdroid <http://code.google.com/p/sipdroid/>*.

--
Rogoshchenkov Nikolay


On Thu, Aug 16, 2012 at 1:58 PM, freeswitch <freeswitch at yeah.net> wrote:

> IMSDROID will be good!
>
>
>
> At 2012-08-16 20:44:54,BookBag <asaad2 at gmail.com> wrote:
>
> I'm surprised no one said Bria by counterpath. Which allows for xmpp and
> presence. The softphone allows you to send sms also if you buy the full
> version. Another one that I tested is Csipsimple, this one allows sms for
> free. But my absolute favorite is Zoiper, unfortunately no sms capability
> but wasnt buggy except for one bug I encountered with sending the correct
> dtmf style. Thank god their new update took care of that. Phone now seems
> very solid and now includes a QR reader.
>
> On Thu, Aug 16, 2012 at 6:16 AM, vaad.fabi at gmail.com <vaad.fabi at gmail.com>wrote:
>
>>   Does they provide these software development, customisation and
>> support ? i think no.  I need vendor\development team who can provide
>> development\customisation\support.
>> Anyway thx.
>>
>>
>> On 08/16/2012 12:53 PM, Mohammad Emran wrote:
>> > Then try nimbuzz or fring.... If it works for u.
>> >
>> > Sent from my iPhone
>> >
>> > On Aug 16, 2012, at 3:42 PM, "vaad.fabi at gmail.com"<vaad.fabi at gmail.com>
>>  wrote:
>> >
>> >>   Seems to be ok for a simple dialer, but i forgot to mention - i need
>> >> advanced options for dialers, such as: IM (xmpp\facebook\twitter
>> >> integration), video calls, balance state, check direction rate,
>> >> callback, sms, top up and etc. It's smth like UC application with
>> >> calling support as one of the many options.
>> >>
>> >>
>> >> On 08/16/2012 12:22 PM, Mohammad Emran wrote:
>> >>> Try www.sipmobiledialer.com
>> >>>
>> >>>
>> >>> Sent from my iPad
>> >>>
>> >>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com"<
>> vaad.fabi at gmail.com>   wrote:
>> >>>
>> >>>>   Hi all,
>> >>>>
>> >>>> Please suggest mobile voip dialer vendor for
>> >>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>> >>>> don't suggest google plz
>> >>>>
>> >>>> --
>> >>>> Best Regards,
>> >>>> Vadim F.
>> >>>>
>> >>>>
>> >>>>
>> _________________________________________________________________________
>> >>>> Professional FreeSWITCH Consulting Services:
>> >>>> consulting at freeswitch.org
>> >>>> http://www.freeswitchsolutions.com
>> >>>>
>> >>>> 
>> >>>> 
>> >>>>
>> >>>> Official FreeSWITCH Sites
>> >>>> http://www.freeswitch.org
>> >>>> http://wiki.freeswitch.org
>> >>>> http://www.cluecon.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
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://wiki.freeswitch.org
>> >>> http://www.cluecon.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
>> >>
>> >> --
>> >> Best Regards,
>> >> Vadim F.
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.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
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.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
>>
>>
>> --
>> Best Regards,
>> Vadim F.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mitch.capper at gmail.com  Thu Aug 16 22:17:08 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 16 Aug 2012 11:17:08 -0700
Subject: [Freeswitch-users] Windows 8 Metro / Windows Phone 8 Freeswitch SIP
	Client Challenge
Message-ID: <CALn233phx81LCUAkpMGAWfsuHzn5Gw=EGNhA=KRtbLnSPA=JWw@mail.gmail.com>

A few users have asked about using freeswitch as a softphone on
Windows 8 metro (or Windows 8 UI or WTF they call it now) or Windows
Phone 8.   If anyone is able to put together a proof of concept
freeswitch working with portaudio (or equiv) at the level they can
place a test call and talk (the number and what have you can be hard
coded in I just need to see it actually work on these platforms)  I
will be happy to write a softphone for either or both those platforms.


~Mitch


From govoiper at gmail.com  Thu Aug 16 22:41:17 2012
From: govoiper at gmail.com (SamyGo)
Date: Thu, 16 Aug 2012 23:41:17 +0500
Subject: [Freeswitch-users] FreeSWITCH Auto Dialer - Open source
In-Reply-To: <502D0CAC.8030905@gmail.com>
References: <CA+aeUL_D-1sgqxvxshU9oqELh66Ck-_kTxesMCCjnv3DgDu-yg@mail.gmail.com>
	<502D0CAC.8030905@gmail.com>
Message-ID: <CAJUJwtg7vDr+MDbTvXJJk4HRAr38CTOYY3mRpWwBui0erBXLeg@mail.gmail.com>

Hi,

Well I'd say to figure out your requirements first. I once got into same
situation as your's and believe me reinventing the wheel (tailor made
solution) for exactly the requirement was the best thing. Atleast I know
I've created something which is doing all that I needed and the time spent
on this was less than a week !!

Its upto you and your requirements to choose the solution.

BR
Sammy Go.
On Aug 16, 2012 8:11 PM, "Abaci" <abaci64 at gmail.com> wrote:

>  http://www.ictdialer.org/
>
> On 8/16/2012 9:53 AM, Obie Brown wrote:
>
> Hi,
>
>  I really want to work with an Open Source dialer and FreeSWITCH but the
> only thing I can find for this is Newfies Dialer which does not have
> features of a Auto Dialer as I need. Also its use for Voice Broadcasting is
> limited.
>
>  My question is. What other Auto Dialers that run on FreeSWITCH are
> there. If any.
>
>
>  Thanks,
> Obie Brown
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Thu Aug 16 23:18:54 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 16 Aug 2012 14:18:54 -0500
Subject: [Freeswitch-users] Profiling FreeSWITCH on Centos
In-Reply-To: <CC527DE9.3C0DA%krice@freeswitch.org>
References: <CAKzWOxURgEuVWtQSSm7NxMBS7Z4c-OmcsrPTGD9Lb6eEj0154g@mail.gmail.com>
	<CC527DE9.3C0DA%krice@freeswitch.org>
Message-ID: <CAKbxfG8MQFn4ZygJ=HUMS258Kv-+oA0F3x=NC3SE5jirx2LibQ@mail.gmail.com>

I don't remember his name now but he came and talked to me at dave and
busters and said he saw unusual activity in the kernel.


On Thu, Aug 16, 2012 at 10:37 AM, Ken Rice <krice at freeswitch.org> wrote:
> It Might have been... Collins can you put me in touch with him offlist?
>
>
>
> On 8/16/12 10:11 AM, "Michael Collins" <msc at freeswitch.org> wrote:
>
> Was it Marcel B. over at Audio Now?
> -MC
>
> On Wed, Aug 15, 2012 at 4:17 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> Hey Guys,
>
> Help me remember. I was speaking with one of you at ClueCon about profiling
> FreeSWITCH on Centos5 and Centos6 so we can compare whats going on. I can
> not remember who that was! If it was you, please contact me on IRC (I?m SwK
> on #freeswitch on freenode), or Email me off list
>
> Thanks
> Ken
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From alex at jajah.com  Thu Aug 16 23:23:32 2012
From: alex at jajah.com (Alex Massover)
Date: Thu, 16 Aug 2012 22:23:32 +0300
Subject: [Freeswitch-users] mod_event_zmq thread safety
Message-ID: <569384504C492C4580E88B5D54DFEAEA30CA0BE3EE@jjex01.jajah.dublin>

Hello,

We're using mod_event_zmq module (slightly modified, we use different socket type and IPC instead of TCP), and experience crashes.

As far as I understood the module it creates ZMQ context and all event sent by using same context.

Events callback is called from different threads, isn't it? And can be a situation when different threads access send() api on this context simultaneously.

ZMQ guide strongly prohibits doing that: " You MUST NOT share ?MQ sockets between threads."

And when I do a mutex around send() there're no more crashes (at least today).

Am I missing something or mod_event_zmq is potentially dangerous?

--
Best Regards,
Alex Massover

From jmesquita at freeswitch.org  Thu Aug 16 23:27:02 2012
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 16 Aug 2012 16:27:02 -0300
Subject: [Freeswitch-users] mod_event_zmq thread safety
In-Reply-To: <569384504C492C4580E88B5D54DFEAEA30CA0BE3EE@jjex01.jajah.dublin>
References: <569384504C492C4580E88B5D54DFEAEA30CA0BE3EE@jjex01.jajah.dublin>
Message-ID: <CABH8QX2bagWAW+=_w=4SXReE+g174S7EMKBQeHnEDH_s2gzJtg@mail.gmail.com>

I do *think* that events are dispatched by different workers on the core
indeed which might be the cause of your problem. ZMQ also has a few asserts
on the core that can be quite disruptive and they were working on getting
it working more properly. Dunno what's the status of that right now.

A core dump would help a lot on any case tho.

Jo?o Mesquita



On Thu, Aug 16, 2012 at 4:23 PM, Alex Massover <alex at jajah.com> wrote:

> Hello,
>
> We're using mod_event_zmq module (slightly modified, we use different
> socket type and IPC instead of TCP), and experience crashes.
>
> As far as I understood the module it creates ZMQ context and all event
> sent by using same context.
>
> Events callback is called from different threads, isn't it? And can be a
> situation when different threads access send() api on this context
> simultaneously.
>
> ZMQ guide strongly prohibits doing that: " You MUST NOT share ?MQ sockets
> between threads."
>
> And when I do a mutex around send() there're no more crashes (at least
> today).
>
> Am I missing something or mod_event_zmq is potentially dangerous?
>
> --
> Best Regards,
> Alex Massover
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From curriegrad2004 at gmail.com  Thu Aug 16 23:49:41 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 16 Aug 2012 12:49:41 -0700
Subject: [Freeswitch-users] Profiling FreeSWITCH on Centos
In-Reply-To: <CAKbxfG8MQFn4ZygJ=HUMS258Kv-+oA0F3x=NC3SE5jirx2LibQ@mail.gmail.com>
References: <CAKzWOxURgEuVWtQSSm7NxMBS7Z4c-OmcsrPTGD9Lb6eEj0154g@mail.gmail.com>
	<CC527DE9.3C0DA%krice@freeswitch.org>
	<CAKbxfG8MQFn4ZygJ=HUMS258Kv-+oA0F3x=NC3SE5jirx2LibQ@mail.gmail.com>
Message-ID: <CAMWiMy_Uq2_H8qorqCweCOu_cC2W6=Ob+cz4p=4GQpW7JHxk9g@mail.gmail.com>

Here's the thing - Red Hat just threw the entire kernel tarball
without separating the patches itself.  (i.e. one huge tarball).

So if the person you're talking to is reporting unusal activity in the
kernel, I would propose that a separate profiling task be done with
CentOS as a chroot under Debian to see if anything is different.

On Thu, Aug 16, 2012 at 12:18 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I don't remember his name now but he came and talked to me at dave and
> busters and said he saw unusual activity in the kernel.
>
>
> On Thu, Aug 16, 2012 at 10:37 AM, Ken Rice <krice at freeswitch.org> wrote:
>> It Might have been... Collins can you put me in touch with him offlist?
>>
>>
>>
>> On 8/16/12 10:11 AM, "Michael Collins" <msc at freeswitch.org> wrote:
>>
>> Was it Marcel B. over at Audio Now?
>> -MC
>>
>> On Wed, Aug 15, 2012 at 4:17 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>> Hey Guys,
>>
>> Help me remember. I was speaking with one of you at ClueCon about profiling
>> FreeSWITCH on Centos5 and Centos6 so we can compare whats going on. I can
>> not remember who that was! If it was you, please contact me on IRC (I?m SwK
>> on #freeswitch on freenode), or Email me off list
>>
>> Thanks
>> Ken
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 alex at jajah.com  Thu Aug 16 23:50:48 2012
From: alex at jajah.com (Alex Massover)
Date: Thu, 16 Aug 2012 22:50:48 +0300
Subject: [Freeswitch-users] mod_event_zmq thread safety
In-Reply-To: <CABH8QX2bagWAW+=_w=4SXReE+g174S7EMKBQeHnEDH_s2gzJtg@mail.gmail.com>
References: <569384504C492C4580E88B5D54DFEAEA30CA0BE3EE@jjex01.jajah.dublin>,
	<CABH8QX2bagWAW+=_w=4SXReE+g174S7EMKBQeHnEDH_s2gzJtg@mail.gmail.com>
Message-ID: <569384504C492C4580E88B5D54DFEAEA30CA0BE3EF@jjex01.jajah.dublin>

Hi,

This is what I understood as well. I'm not able to get a core for some reason, but provided that event callback is called by different threads, it strongly suggests that current implementation of mod_event_zmq is unsafe as sharing zmq context is not thread-safe.

What is the process of creating a ticket, please? Should I sing-up to JIRA?

--
Alex.
________________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita [jmesquita at freeswitch.org]
Sent: Thursday, August 16, 2012 10:27 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_event_zmq thread safety

I do *think* that events are dispatched by different workers on the core indeed which might be the cause of your problem. ZMQ also has a few asserts on the core that can be quite disruptive and they were working on getting it working more properly. Dunno what's the status of that right now.

A core dump would help a lot on any case tho.

Jo?o Mesquita



On Thu, Aug 16, 2012 at 4:23 PM, Alex Massover <alex at jajah.com<mailto:alex at jajah.com>> wrote:
Hello,

We're using mod_event_zmq module (slightly modified, we use different socket type and IPC instead of TCP), and experience crashes.

As far as I understood the module it creates ZMQ context and all event sent by using same context.

Events callback is called from different threads, isn't it? And can be a situation when different threads access send() api on this context simultaneously.

ZMQ guide strongly prohibits doing that: " You MUST NOT share ?MQ sockets between threads."

And when I do a mutex around send() there're no more crashes (at least today).

Am I missing something or mod_event_zmq is potentially dangerous?

--
Best Regards,
Alex Massover
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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



From msc at freeswitch.org  Thu Aug 16 23:52:09 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Aug 2012 12:52:09 -0700
Subject: [Freeswitch-users] Profiling FreeSWITCH on Centos
In-Reply-To: <CAKbxfG8MQFn4ZygJ=HUMS258Kv-+oA0F3x=NC3SE5jirx2LibQ@mail.gmail.com>
References: <CAKzWOxURgEuVWtQSSm7NxMBS7Z4c-OmcsrPTGD9Lb6eEj0154g@mail.gmail.com>
	<CC527DE9.3C0DA%krice@freeswitch.org>
	<CAKbxfG8MQFn4ZygJ=HUMS258Kv-+oA0F3x=NC3SE5jirx2LibQ@mail.gmail.com>
Message-ID: <CAKzWOxXWuyADQqhcudw0mOzLJvTvEjbjzz5ar9agEisaj2Kz_g@mail.gmail.com>

That sounds like Marcel! I've sent his info over to Ken. If anyone else has
any information on CentOS 6 and FS then please get in touch with Ken.

-MC

On Thu, Aug 16, 2012 at 12:18 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I don't remember his name now but he came and talked to me at dave and
> busters and said he saw unusual activity in the kernel.
>
>
> On Thu, Aug 16, 2012 at 10:37 AM, Ken Rice <krice at freeswitch.org> wrote:
> > It Might have been... Collins can you put me in touch with him offlist?
> >
> >
> >
> > On 8/16/12 10:11 AM, "Michael Collins" <msc at freeswitch.org> wrote:
> >
> > Was it Marcel B. over at Audio Now?
> > -MC
> >
> > On Wed, Aug 15, 2012 at 4:17 PM, Ken Rice <krice at freeswitch.org> wrote:
> >
> > Hey Guys,
> >
> > Help me remember. I was speaking with one of you at ClueCon about
> profiling
> > FreeSWITCH on Centos5 and Centos6 so we can compare whats going on. I can
> > not remember who that was! If it was you, please contact me on IRC (I?m
> SwK
> > on #freeswitch on freenode), or Email me off list
> >
> > Thanks
> > Ken
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From ga at steadfasttelecom.com  Fri Aug 17 00:15:54 2012
From: ga at steadfasttelecom.com (Gilad Abada)
Date: Thu, 16 Aug 2012 16:15:54 -0400
Subject: [Freeswitch-users] Any good Canadian SIP carriers
In-Reply-To: <C69FF59885AC464896A4B68500B6E974098CF5E2@VAL-SV-EXCH-01.valiant.com>
References: <C69FF59885AC464896A4B68500B6E974098CF5E2@VAL-SV-EXCH-01.valiant.com>
Message-ID: <CAKD7_2T5dbv-YCKhRHsbxiEn_BTkkw494dMpM3nu6JnsAgBrFA@mail.gmail.com>

Go with Flowroute.

Their prices are amazing and they have Canada origination and termination.

On Tue, Aug 14, 2012 at 4:08 PM, Phil Kim <phil.kim at valiant.com> wrote:

> ** **
>
> Hello all,****
>
> ** **
>
> I am looking for a good SIP carrier (provider) in ****Canada****. We
> already have a carrier in **U.S.** but they do not offer good rates for
> calls coming over from ****Canada****. ****
>
> ** **
>
> Any inputs will be appreciated. ****
>
> ** **
>
> Thank you.****
>
> ** **
>
> Best regards,****
>
> ** **
>
> Phil H. Kim****
>
> ** **
>
> The information contained in this e-mail message, and any attachment
> thereto, is the property of Valiant and is confidential and may not be
> disclosed without our express permission. If you are not the intended
> recipient or an employee or agent responsible for delivering this message
> to
> the intended recipient, you are hereby notified that you have received this
> message in error and that any review, dissemination, distribution or
> copying
> of this message, or any attachment thereto, in whole or in part, is
> strictly
> prohibited. If you have received this message in error, please immediately
> notify us by telephone, fax or e-mail and delete the message and all of its
> attachments.
>
> Thank you.
>
> Every effort is made to keep our network free from viruses. You should,
> however, review this e-mail message, as well as any attachment thereto,
> for viruses. We take no responsibility and have no liability for any
> computer virus which may be transferred via this e-mail message.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Gilad Abada

SteadFast Telecommunications, Inc.

Call us to find out how much you can save with VoIP!

V: 212.589.1001
F: 212.589.1011


For 35 years, Steadfast Telecommunications has been providing
state-of-the-art communications technology to businesses and government
agencies - large and small. Steadfast Telecommunications tailors Unified
Communications and Voice-Over IP Solutions to single-site offices or
multi-site and worldwide enterprises.   Make your virtual office a
reality.  Enjoy the freedom to travel while remaining connected to your
office.
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From msc at freeswitch.org  Fri Aug 17 00:48:13 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 16 Aug 2012 13:48:13 -0700
Subject: [Freeswitch-users] Any good Canadian SIP carriers
In-Reply-To: <CAKD7_2T5dbv-YCKhRHsbxiEn_BTkkw494dMpM3nu6JnsAgBrFA@mail.gmail.com>
References: <C69FF59885AC464896A4B68500B6E974098CF5E2@VAL-SV-EXCH-01.valiant.com>
	<CAKD7_2T5dbv-YCKhRHsbxiEn_BTkkw494dMpM3nu6JnsAgBrFA@mail.gmail.com>
Message-ID: <CAKzWOxUcyjyS5SqvhvnfBuKULrevOz6h6PR7CP6=FnEsDNps7A@mail.gmail.com>

Plus, they're really cool people and they were Gold sponsors of ClueCon.
Tell them the FreeSWITCH community sent you and that you are happy they
support OSS telephony. :)

-MC

On Thu, Aug 16, 2012 at 1:15 PM, Gilad Abada <ga at steadfasttelecom.com>wrote:

> Go with Flowroute.
>
> Their prices are amazing and they have Canada origination and termination.
>
> On Tue, Aug 14, 2012 at 4:08 PM, Phil Kim <phil.kim at valiant.com> wrote:
>
>> ** **
>>
>> Hello all,****
>>
>> ** **
>>
>> I am looking for a good SIP carrier (provider) in ****Canada****. We
>> already have a carrier in **U.S.** but they do not offer good rates for
>> calls coming over from ****Canada****. ****
>>
>> ** **
>>
>> Any inputs will be appreciated. ****
>>
>> ** **
>>
>> Thank you.****
>>
>> ** **
>>
>> Best regards,****
>>
>> ** **
>>
>> Phil H. Kim****
>>
>> ** **
>>
>> The information contained in this e-mail message, and any attachment
>> thereto, is the property of Valiant and is confidential and may not be
>> disclosed without our express permission. If you are not the intended
>> recipient or an employee or agent responsible for delivering this message
>> to
>> the intended recipient, you are hereby notified that you have received
>> this
>> message in error and that any review, dissemination, distribution or
>> copying
>> of this message, or any attachment thereto, in whole or in part, is
>> strictly
>> prohibited. If you have received this message in error, please immediately
>> notify us by telephone, fax or e-mail and delete the message and all of
>> its
>> attachments.
>>
>> Thank you.
>>
>> Every effort is made to keep our network free from viruses. You should,
>> however, review this e-mail message, as well as any attachment thereto,
>> for viruses. We take no responsibility and have no liability for any
>> computer virus which may be transferred via this e-mail message.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Gilad Abada
>
> SteadFast Telecommunications, Inc.
>
> Call us to find out how much you can save with VoIP!
>
> V: 212.589.1001
> F: 212.589.1011
>
>
> For 35 years, Steadfast Telecommunications has been providing
> state-of-the-art communications technology to businesses and government
> agencies - large and small. Steadfast Telecommunications tailors Unified
> Communications and Voice-Over IP Solutions to single-site offices or
> multi-site and worldwide enterprises.   Make your virtual office a
> reality.  Enjoy the freedom to travel while remaining connected to your
> office.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From krice at freeswitch.org  Fri Aug 17 01:37:42 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 16 Aug 2012 16:37:42 -0500
Subject: [Freeswitch-users] Profiling FreeSWITCH on Centos
In-Reply-To: <CAKzWOxXWuyADQqhcudw0mOzLJvTvEjbjzz5ar9agEisaj2Kz_g@mail.gmail.com>
Message-ID: <CC52D256.3C0FE%krice@freeswitch.org>

Yes Please, we would like to figure out what it is and if reasonable work
around it, or atleast, upstream the issue to see if we can get Centos to
resolve it...

Also, if you have a RHEL6 box I can test on please contact me offlist so we
can arrange for this. Currently we are having really good luck with all the
other platforms we support lets not leave out a widely used OS

K


On 8/16/12 2:52 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> That sounds like Marcel! I've sent his info over to Ken. If anyone else has
> any information on CentOS 6 and FS then please get in touch with Ken.
> 
> -MC
> 
> On Thu, Aug 16, 2012 at 12:18 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> I don't remember his name now but he came and talked to me at dave and
>> busters and said he saw unusual activity in the kernel.
>> 
>> 
>> On Thu, Aug 16, 2012 at 10:37 AM, Ken Rice <krice at freeswitch.org> wrote:
>>> > It Might have been... Collins can you put me in touch with him offlist?
>>> >
>>> >
>>> >
>>> > On 8/16/12 10:11 AM, "Michael Collins" <msc at freeswitch.org> wrote:
>>> >
>>> > Was it Marcel B. over at Audio Now?
>>> > -MC
>>> >
>>> > On Wed, Aug 15, 2012 at 4:17 PM, Ken Rice <krice at freeswitch.org> wrote:
>>> >
>>> > Hey Guys,
>>> >
>>> > Help me remember. I was speaking with one of you at ClueCon about
>>> profiling
>>> > FreeSWITCH on Centos5 and Centos6 so we can compare whats going on. I can
>>> > not remember who that was! If it was you, please contact me on IRC (I?m
>>> SwK
>>> > on #freeswitch on freenode), or Email me off list
>>> >
>>> > Thanks
>>> > Ken
>>> >
>>> > _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.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
>>> >
>>> >
>>> >
>>> >
>>> > _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>> 
>> 
>> 
>> --
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> <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 <tel:%2B19193869900>
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 alex at thewinelake.com  Fri Aug 17 01:44:14 2012
From: alex at thewinelake.com (Alexander Lake)
Date: Thu, 16 Aug 2012 22:44:14 +0100
Subject: [Freeswitch-users] CALL TIMEOUT and interprise bridge
In-Reply-To: <00a201cd7bd5$9382dd50$ba8897f0$@bizfocused.com>
References: <00a201cd7bd5$9382dd50$ba8897f0$@bizfocused.com>
Message-ID: <0723543F-CE11-4C85-81D8-4B13E633DC74@thewinelake.com>

You can try leg_timeout.

On 16 Aug 2012, at 18:35, Sean Devoy wrote:

> Hi,
> I have noticed that ?SET [call_timeout]=[25]? seems to be ignored when using an Enterprise bridge (:_:) to two extensions.  The timeout is always 60 seconds.
>  
> Do I just need to set the timeout in {} on each leg?
>  
> Sean
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 brett at launch3.net  Fri Aug 17 02:28:12 2012
From: brett at launch3.net (Brett Wilson)
Date: Thu, 16 Aug 2012 18:28:12 -0400
Subject: [Freeswitch-users] Dropping calls after 40 secs
In-Reply-To: <CAKV-jFw5mrzxag-QeuCBL0=NRbH7FonPajshNuWEth6=waGNRg@mail.gmail.com>
References: <cbb3a1a7-d000-4a3c-81d2-cfc7c600bce5.maildroid@localhost>	<CANBhyPANkTy5YEe2uCbwFCGSCqFELjKt9H4_D8-rs3yB-LAG0g@mail.gmail.com>	<005801cd7bc4$3e394700$baabd500$@launch3.net>
	<CAKV-jFw5mrzxag-QeuCBL0=NRbH7FonPajshNuWEth6=waGNRg@mail.gmail.com>
Message-ID: <012a01cd7bfe$6c6c9f20$4545dd60$@launch3.net>

Here you go:

http://pastebin.com/RKtDCA4N

 

******************************************* 
Brett Wilson
IT Department
Launch 3 Ventures, LLC 
134 Myer Street 
Hackensack, NJ 07601 
Phone: 877.878.9134
Fax: 646.536.3866 
Email: Brett.Wilson at launch3.net
AOL IM: Brett.Wilson at launch3.net
 <http://www.launch3.net/> www.Launch3.net 
www.Launch3telecom.com <http://www.launch3telecom.com/>  
******************************************* 
 <http://launch3telecom.blogspot.com/> Description: Description:
Description: Blogger-logo
<http://www.facebook.com/pages/Launch-3-Telecom/165610820178720>
Description: Description: Description: FaceBook-Logo
<https://twitter.com/launch3telecom> Description: Description: Description:
Twitter-Logo <https://plus.google.com/102713212642017290959> Description:
Description: Description: GPlus-Logo

 

From: Brian Foster [mailto:bdfoster at endigotech.com] 
Sent: Thursday, August 16, 2012 12:52 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping calls after 40 secs

 

Need the siptrace of all legs involved, combined with the debug that you
posted. We need to be able to look at the sip messages involved to find
clues. 

sofia profile internal siptrace on
sofia profile external siptrace on

Brian Foster
Endigo Computer LLC

Sent from a mobile device.

On Aug 16, 2012 11:36 AM, "Brett Wilson" <brett at launch3.net> wrote:

What do you need logs of? I posted that pastebin link for the log output on
max verbosity. I do have auto-nat enabled.

 

******************************************* 
Brett Wilson
IT Department
Launch 3 Ventures, LLC 
134 Myer Street 
Hackensack, NJ 07601 
Phone: 877.878.9134
Fax: 646.536.3866 
Email: Brett.Wilson at launch3.net
AOL IM: Brett.Wilson at launch3.net
 <http://www.launch3.net/> www.Launch3.net 
www.Launch3telecom.com <http://www.launch3telecom.com/>  
******************************************* 
 <http://launch3telecom.blogspot.com/> Description: Description:
Description: Blogger-logo
<http://www.facebook.com/pages/Launch-3-Telecom/165610820178720>
Description: Description: Description: FaceBook-Logo
<https://twitter.com/launch3telecom> Description: Description: Description:
Twitter-Logo <https://plus.google.com/102713212642017290959> Description:
Description: Description: GPlus-Logo

 

From: Anton Kvashenkin [mailto:anton.jugatsu at gmail.com] 
Sent: Thursday, August 16, 2012 10:20 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropping calls after 40 secs

 

Must be NAT, but we can not tell you exactly cause you didn't provide any
logs.

16.08.2012 18:11 ???????????? "Brett Wilson" <brett at launch3.net> ???????:

hello,
I am having issues with calls being dropped after about 40 secs. I just
compiled and installed the latest fs. Here is log output:
pastebin.com/abJJfLUi 

******************************************* 
Brett Wilson 
IT Department
Launch 3 Ventures, LLC 
134 Myer Street 
Hackensack, NJ 07601 
Phone: 877.878.9134 <tel:8778789134>  
Fax: 646.536.3866 <tel:6465363866>  
Cell: 973.650.7189 <tel:9736507189> 
Email: brett at launch3.net 
www.Launch3.net 
www.Launch3telecom.com *******************************************


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 dujinfang at gmail.com  Fri Aug 17 04:36:29 2012
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 17 Aug 2012 08:36:29 +0800
Subject: [Freeswitch-users] 3 sessions instead of one. with use loopback.
In-Reply-To: <CANJXrA-ZaKcSvmpvJJOMgouovD-08NmGe8zDxO2R_JY5d7GvLw@mail.gmail.com>
References: <CANJXrA-ZaKcSvmpvJJOMgouovD-08NmGe8zDxO2R_JY5d7GvLw@mail.gmail.com>
Message-ID: <EE305D5045EC4E3C9E14DE27236C818A@gmail.com>

try http://wiki.freeswitch.org/wiki/Variable_loopback_bowout_on_execute 

-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Thursday, August 16, 2012 at 8:29 PM, Prevail Magid wrote:

> Hi,
> 
> I have ESL client that send comand to FS for invite person to conference like : "conference 777 dial loopback/ADD_CONF_1007"
> 
> and dialplan for catch this :
> 
>        <condition field="destination_number" expression="^ADD_CONF_(\d{4})$" break="on-true">
>                         <action application="set" data="hangup_after_bridge=true" />
> <action application="bridge" data="user/$1" />
> </condition>
> 
> 
> and I have three channels :  loopback/ADD_CONF-a and loopback/ADD_CONF-b - (that`s created loopback?) and sofia/internal/sip... 
> 
> How can I close loopback channels, thats do not need as understand after bridge? 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> 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
> 
> 
> 


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From sameer2k3t at gmail.com  Fri Aug 17 06:52:02 2012
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Fri, 17 Aug 2012 07:52:02 +0500
Subject: [Freeswitch-users] identify ttyUSB for gsmopen
Message-ID: <CAA2gtHCD8s03R6Pux_rp6HWuH9VmBkCRVANObCq_TYnote2Z0A@mail.gmail.com>

Hi,

This is just to share how we can find out  which serial to be used with
gsmopen


Install wvdial :

yum install wvdial

then run :

wvdialconf /etc/wvdial.conf

It will display output like :

Editing `/etc/wvdial.conf'.

Scanning your serial ports for a modem.

ttyS0<*1>: ATQ0 V1 E1 -- failed with 2400 baud, next try: 9600 baud
ttyS0<*1>: ATQ0 V1 E1 -- failed with 9600 baud, next try: 115200 baud
ttyS0<*1>: ATQ0 V1 E1 -- and failed too at 115200, giving up.
ttyS1<*1>: ATQ0 V1 E1 -- failed with 2400 baud, next try: 9600 baud
ttyS1<*1>: ATQ0 V1 E1 -- failed with 9600 baud, next try: 115200 baud
ttyS1<*1>: ATQ0 V1 E1 -- and failed too at 115200, giving up.
Modem Port Scan<*1>: S2   S3
WvModem<*1>: Cannot get information for serial port.
ttyUSB3<*1>: ATQ0 V1 E1 -- OK
ttyUSB3<*1>: ATQ0 V1 E1 Z -- OK
ttyUSB3<*1>: ATQ0 V1 E1 S0=0 -- OK
ttyUSB3<*1>: ATQ0 V1 E1 S0=0 &C1 -- OK
ttyUSB3<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 -- OK
ttyUSB3<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0 -- OK
ttyUSB3<*1>: Modem Identifier: ATI -- Manufacturer: huawei
ttyUSB3<*1>: Speed 9600: AT -- OK
ttyUSB3<*1>: Max speed is 9600; that should be safe.
ttyUSB3<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0 -- OK
WvModem<*1>: Cannot get information for serial port.
ttyUSB4<*1>: ATQ0 V1 E1 -- failed with 2400 baud, next try: 9600 baud
ttyUSB4<*1>: ATQ0 V1 E1 -- failed with 9600 baud, next try: 9600 baud
ttyUSB4<*1>: ATQ0 V1 E1 -- and failed too at 115200, giving up.
WvModem<*1>: Cannot get information for serial port.
ttyUSB5<*1>: ATQ0 V1 E1 -- failed with 2400 baud, next try: 9600 baud
ttyUSB5<*1>: ATQ0 V1 E1 -- failed with 9600 baud, next try: 9600 baud
ttyUSB5<*1>: ATQ0 V1 E1 -- and failed too at 115200, giving up.
WvModem<*1>: Cannot get information for serial port.
ttyUSB6<*1>: ATQ0 V1 E1 -- OK
ttyUSB6<*1>: ATQ0 V1 E1 Z -- OK
ttyUSB6<*1>: ATQ0 V1 E1 S0=0 -- OK
ttyUSB6<*1>: ATQ0 V1 E1 S0=0 &C1 -- OK
ttyUSB6<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 -- OK
ttyUSB6<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0 -- OK
ttyUSB6<*1>: Modem Identifier: ATI -- Manufacturer: huawei
ttyUSB6<*1>: Speed 9600: AT -- OK
ttyUSB6<*1>: Max speed is 9600; that should be safe.
ttyUSB6<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0 -- OK

Found a modem on /dev/ttyUSB3.
Modem configuration written to /etc/wvdial.conf.
ttyUSB3<Info>: Speed 9600; init "ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0"
ttyUSB6<Info>: Speed 9600; init "ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0"
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From ntomer at newgen.co.in  Fri Aug 17 08:06:50 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Fri, 17 Aug 2012 09:36:50 +0530
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
Message-ID: <1345176410.502dc35a7c3ec@mx.newgen.co.in>

Please help.

On Thursday, 16-08-2012 on 17:10 Nitin Tomer wrote:

Hi,

?

I am working on a Contact Center solution. It will support mail, chat
and call queries.

The requirements are:

1. An end-customer calls, the call is handled by FreeSWITCH Auto
Attendant.
2. Customer is presented with a menu and makes selection. His call is
put on hold and an entry is made in my system's database for incoming
queries.
3. These queries are shown to agents handling calls.
4. And agent clicks on a query, he is shown an extension where call
is parked. He dials that and is connected to the customer.

5. He talks to the customer and resolve his queries.

Please guide me on how can do it. The application will be written in
Java.

I am an experienced programmers in Java/J2EE; but doesn't have much
knowledge of VoIP/FreeSWITCH. I?ve configured FreeSWITCH on Ubuntu
12.04; and have installed X-Lite Softphones on two Windows machines. I
configured these phones to work with FreeSWITCH and they are working
fine.

?

I think that I?d have to use Valet Parking to park customer?s
call on an extension, then pass that extension to agent who will dial
it and will be connected to the customer.

?

Please tell me whether this approach will work? And how should I go
about it.

?

All help would be much appreciated.

?

Regards

?

Nitin


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From gabe at gundy.org  Fri Aug 17 10:58:23 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 17 Aug 2012 00:58:23 -0600
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <1345176410.502dc35a7c3ec@mx.newgen.co.in>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<1345176410.502dc35a7c3ec@mx.newgen.co.in>
Message-ID: <CAMwZ-t+KUp494GT90dV7oS02sviMVerxJBNCVtyRosg_qneBRA@mail.gmail.com>

On Thu, Aug 16, 2012 at 10:06 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:
> Please help.

You're going to need to be more patient than that. You let 17 full
hours pass. You need to give people time to sleep and do their day
jobs.

Anyway, good luck.


Gabe


From vaad.fabi at gmail.com  Fri Aug 17 11:05:15 2012
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Fri, 17 Aug 2012 10:05:15 +0300
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <7357444f.60.1393095b1e5.Coremail.freeswitch@yeah.net>
References: <4DB00624.2010009@tagnet.ru>
	<502CBA3E.7030107@gmail.com>	<86E0D1C3-8B78-4A71-9BA4-1C901B40FCDD@gmail.com>	<502CC06B.8070202@gmail.com>	<806E15A7-FD82-4215-B50E-9B921CB2DC7D@gmail.com>	<502CD69D.1060908@gmail.com>	<CAAdTMa9=+4bQtJ75NAR+uDckFW8Q531iMSwWo17nYjbx9co5CQ@mail.gmail.com>
	<7357444f.60.1393095b1e5.Coremail.freeswitch@yeah.net>
Message-ID: <502DED2B.8090105@gmail.com>


IMSDROID - only Android platform, no centralizied dev team\vendor with 
support and SLA
SIPDROID - only Android platform, no centralizied dev team\vendor with 
support and SLA
Counterpath - as i know no mobile platforms

Searching...


On 08/16/2012 08:58 PM, freeswitch wrote:
> IMSDROID will be good!
>
>
>
> At 2012-08-16 20:44:54,BookBag <asaad2 at gmail.com> wrote:
>
>     I'm surprised no one said Bria by counterpath. Which allows for
>     xmpp and presence. The softphone allows you to send sms also if
>     you buy the full version. Another one that I tested is Csipsimple,
>     this one allows sms for free. But my absolute favorite is Zoiper,
>     unfortunately no sms capability but wasnt buggy except for one bug
>     I encountered with sending the correct dtmf style. Thank god their
>     new update took care of that. Phone now seems very solid and now
>     includes a QR reader.
>
>     On Thu, Aug 16, 2012 at 6:16 AM, vaad.fabi at gmail.com
>     <mailto:vaad.fabi at gmail.com> <vaad.fabi at gmail.com
>     <mailto:vaad.fabi at gmail.com>> wrote:
>
>         Does they provide these software development, customisation and
>         support ? i think no. I need vendor\development team who can
>         provide
>         development\customisation\support.
>         Anyway thx.
>
>
>         On 08/16/2012 12:53 PM, Mohammad Emran wrote:
>         > Then try nimbuzz or fring.... If it works for u.
>         >
>         > Sent from my iPhone
>         >
>         > On Aug 16, 2012, at 3:42 PM, "vaad.fabi at gmail.com
>         <mailto:vaad.fabi at gmail.com>"<vaad.fabi at gmail.com
>         <mailto:vaad.fabi at gmail.com>> wrote:
>         >
>         >> Seems to be ok for a simple dialer, but i forgot to mention
>         - i need
>         >> advanced options for dialers, such as: IM
>         (xmpp\facebook\twitter
>         >> integration), video calls, balance state, check direction rate,
>         >> callback, sms, top up and etc. It's smth like UC
>         application with
>         >> calling support as one of the many options.
>         >>
>         >>
>         >> On 08/16/2012 12:22 PM, Mohammad Emran wrote:
>         >>> Try www.sipmobiledialer.com <http://www.sipmobiledialer.com>
>         >>>
>         >>>
>         >>> Sent from my iPad
>         >>>
>         >>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com
>         <mailto:vaad.fabi at gmail.com>"<vaad.fabi at gmail.com
>         <mailto:vaad.fabi at gmail.com>> wrote:
>         >>>
>         >>>> Hi all,
>         >>>>
>         >>>> Please suggest mobile voip dialer vendor for
>         >>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>         >>>> don't suggest google plz
>         >>>>
>         >>>> --
>         >>>> Best Regards,
>         >>>> Vadim F.
>         >>>>
>         >>>>
>         >>>>
>         _________________________________________________________________________
>         >>>> Professional FreeSWITCH Consulting Services:
>         >>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         >>>> http://www.freeswitchsolutions.com
>         >>>>
>         >>>> 
>         >>>> 
>         >>>>
>         >>>> Official FreeSWITCH Sites
>         >>>> http://www.freeswitch.org
>         >>>> http://wiki.freeswitch.org
>         >>>> 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
>         >>>
>         _________________________________________________________________________
>         >>> Professional FreeSWITCH Consulting Services:
>         >>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         >>> http://www.freeswitchsolutions.com
>         >>>
>         >>> 
>         >>> 
>         >>>
>         >>> Official FreeSWITCH Sites
>         >>> http://www.freeswitch.org
>         >>> http://wiki.freeswitch.org
>         >>> 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
>         >>
>         >> --
>         >> Best Regards,
>         >> Vadim F.
>         >>
>         >>
>         >>
>         _________________________________________________________________________
>         >> Professional FreeSWITCH Consulting Services:
>         >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         >> http://www.freeswitchsolutions.com
>         >>
>         >> 
>         >> 
>         >>
>         >> Official FreeSWITCH Sites
>         >> http://www.freeswitch.org
>         >> http://wiki.freeswitch.org
>         >> 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
>         >
>         _________________________________________________________________________
>         > Professional FreeSWITCH Consulting Services:
>         > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         > http://www.freeswitchsolutions.com
>         >
>         > 
>         > 
>         >
>         > Official FreeSWITCH Sites
>         > http://www.freeswitch.org
>         > http://wiki.freeswitch.org
>         > 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
>
>
>         --
>         Best Regards,
>         Vadim F.
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         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
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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


-- 
Best Regards,
Vadim F.

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From itamar at ispbrasil.com.br  Fri Aug 17 11:13:04 2012
From: itamar at ispbrasil.com.br (Itamar Reis Peixoto)
Date: Fri, 17 Aug 2012 04:13:04 -0300
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <502DED2B.8090105@gmail.com>
References: <4DB00624.2010009@tagnet.ru> <502CBA3E.7030107@gmail.com>
	<86E0D1C3-8B78-4A71-9BA4-1C901B40FCDD@gmail.com>
	<502CC06B.8070202@gmail.com>
	<806E15A7-FD82-4215-B50E-9B921CB2DC7D@gmail.com>
	<502CD69D.1060908@gmail.com>
	<CAAdTMa9=+4bQtJ75NAR+uDckFW8Q531iMSwWo17nYjbx9co5CQ@mail.gmail.com>
	<7357444f.60.1393095b1e5.Coremail.freeswitch@yeah.net>
	<502DED2B.8090105@gmail.com>
Message-ID: <CACyzUwKEbfQT15DBjfJ8TvcNDVotFEXcwpW44RnX9+M_Emwtbg@mail.gmail.com>

On Fri, Aug 17, 2012 at 4:05 AM, vaad.fabi at gmail.com
<vaad.fabi at gmail.com> wrote:
>
> IMSDROID - only Android platform, no  centralizied dev team\vendor with
> support and SLA
> SIPDROID - only Android platform, no  centralizied dev team\vendor with
> support and SLA
> Counterpath - as i know no mobile platforms
>
> Searching...
>


Android sip client -> settings -> call settings -> accounts ->

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

Itamar Reis Peixoto


From shaheryarkh at googlemail.com  Fri Aug 17 11:45:46 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Fri, 17 Aug 2012 09:45:46 +0200
Subject: [Freeswitch-users] mobile dialers
In-Reply-To: <502DED2B.8090105@gmail.com>
References: <4DB00624.2010009@tagnet.ru> <502CBA3E.7030107@gmail.com>
	<86E0D1C3-8B78-4A71-9BA4-1C901B40FCDD@gmail.com>
	<502CC06B.8070202@gmail.com>
	<806E15A7-FD82-4215-B50E-9B921CB2DC7D@gmail.com>
	<502CD69D.1060908@gmail.com>
	<CAAdTMa9=+4bQtJ75NAR+uDckFW8Q531iMSwWo17nYjbx9co5CQ@mail.gmail.com>
	<7357444f.60.1393095b1e5.Coremail.freeswitch@yeah.net>
	<502DED2B.8090105@gmail.com>
Message-ID: <CAFZQphxVNSXBA010cFGdkTi-ELfYyJ89xmBQ+DughScSAAF0JQ@mail.gmail.com>

IMSDroid is based on Doubango developed by Doubango Telecom, which also has
support for iPhone, Windows and Mac.

You should check their website at http://www.doubango.org. I think they may
provide you commercial support for their productions as well, just talk to
them.

Thank you.


On Fri, Aug 17, 2012 at 9:05 AM, vaad.fabi at gmail.com <vaad.fabi at gmail.com>wrote:

> **
>
> IMSDROID - only Android platform, no  centralizied dev team\vendor with
> support and SLA
> SIPDROID - only Android platform, no  centralizied dev team\vendor with
> support and SLA
> Counterpath - as i know no mobile platforms
>
> Searching...
>
>
>
> On 08/16/2012 08:58 PM, freeswitch wrote:
>
>  IMSDROID will be good!
>
>
>
>  At 2012-08-16 20:44:54,BookBag <asaad2 at gmail.com> <asaad2 at gmail.com>wrote:
>
> I'm surprised no one said Bria by counterpath. Which allows for xmpp and
> presence. The softphone allows you to send sms also if you buy the full
> version. Another one that I tested is Csipsimple, this one allows sms for
> free. But my absolute favorite is Zoiper, unfortunately no sms capability
> but wasnt buggy except for one bug I encountered with sending the correct
> dtmf style. Thank god their new update took care of that. Phone now seems
> very solid and now includes a QR reader.
>
> On Thu, Aug 16, 2012 at 6:16 AM, vaad.fabi at gmail.com <vaad.fabi at gmail.com>wrote:
>
>>   Does they provide these software development, customisation and
>> support ? i think no.  I need vendor\development team who can provide
>> development\customisation\support.
>> Anyway thx.
>>
>>
>> On 08/16/2012 12:53 PM, Mohammad Emran wrote:
>> > Then try nimbuzz or fring.... If it works for u.
>> >
>> > Sent from my iPhone
>> >
>> > On Aug 16, 2012, at 3:42 PM, "vaad.fabi at gmail.com"<vaad.fabi at gmail.com>
>>  wrote:
>> >
>> >>   Seems to be ok for a simple dialer, but i forgot to mention - i need
>> >> advanced options for dialers, such as: IM (xmpp\facebook\twitter
>> >> integration), video calls, balance state, check direction rate,
>> >> callback, sms, top up and etc. It's smth like UC application with
>> >> calling support as one of the many options.
>> >>
>> >>
>> >> On 08/16/2012 12:22 PM, Mohammad Emran wrote:
>> >>> Try www.sipmobiledialer.com
>> >>>
>> >>>
>> >>> Sent from my iPad
>> >>>
>> >>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com"<
>> vaad.fabi at gmail.com>   wrote:
>> >>>
>> >>>>   Hi all,
>> >>>>
>> >>>> Please suggest mobile voip dialer vendor for
>> >>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>> >>>> don't suggest google plz
>> >>>>
>> >>>> --
>> >>>> Best Regards,
>> >>>> Vadim F.
>> >>>>
>> >>>>
>> >>>>
>> _________________________________________________________________________
>> >>>> Professional FreeSWITCH Consulting Services:
>> >>>> consulting at freeswitch.org
>> >>>> http://www.freeswitchsolutions.com
>> >>>>
>> >>>> 
>> >>>> 
>> >>>>
>> >>>> Official FreeSWITCH Sites
>> >>>> http://www.freeswitch.org
>> >>>> http://wiki.freeswitch.org
>> >>>> http://www.cluecon.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
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://wiki.freeswitch.org
>> >>> http://www.cluecon.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
>> >>
>> >> --
>> >> Best Regards,
>> >> Vadim F.
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.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
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.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
>>
>>
>> --
>> Best Regards,
>> Vadim F.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
>
>
> --
> Best Regards,
> Vadim F.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From basit.engg at gmail.com  Fri Aug 17 12:14:20 2012
From: basit.engg at gmail.com (Abdul Basit)
Date: Fri, 17 Aug 2012 13:14:20 +0500
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <CAMwZ-t+KUp494GT90dV7oS02sviMVerxJBNCVtyRosg_qneBRA@mail.gmail.com>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<1345176410.502dc35a7c3ec@mx.newgen.co.in>
	<CAMwZ-t+KUp494GT90dV7oS02sviMVerxJBNCVtyRosg_qneBRA@mail.gmail.com>
Message-ID: <CAGQG5MZFkiSnJhJ6iwuGrYSgtcj5N1JCPatDpVH0rYFg9kSM5w@mail.gmail.com>

You can consider mod_callcenter

http://www.thenoccave.com/2011/10/17/freeswitch-queues-with-mod_callcenter/

Implement call queues that deal with ACD.

Collect information from client using autoattendent system, insert queries
into db and pass call to Queue.

Attach few agents to that Queue where any free agent will get this call.

Using CTI-popup populate client preferences to the selected agent.

I hope this will help.

--
regards,

abdul basit


On Fri, Aug 17, 2012 at 11:58 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Thu, Aug 16, 2012 at 10:06 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:
> > Please help.
>
> You're going to need to be more patient than that. You let 17 full
> hours pass. You need to give people time to sleep and do their day
> jobs.
>
> Anyway, good luck.
>
>
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 clive18 at webmail.co.za  Fri Aug 17 12:24:49 2012
From: clive18 at webmail.co.za (clive engelberg)
Date: Fri, 17 Aug 2012 10:24:49 +0200
Subject: [Freeswitch-users] mobile diallers
In-Reply-To: <mailman.17045.1345189578.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.17045.1345189578.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <26f073599ffd326824ba44e1ec40fa4f@www.webmail.co.za>

Hi 

 Someone recommended http://www.acrobits.cz 

 Worth a try. 

 regards 

 Clive 

 On Fri, 17 Aug 2012 11:46:18 +0400
freeswitch-users-request at lists.freeswitch.org wrote

  IMSDroid is based on Doubango developed by Doubango Telecom, which also has
support for iPhone, Windows and Mac.
 You should check their website at http://www.doubango.org [1]. I think they
may provide you commercial support for their productions as well, just talk
to them.
 Thank you. 

 On Fri, Aug 17, 2012 at 9:05 AM, vaad.fabi at gmail.com [2]  wrote:

 IMSDROID - only Android platform, no centralizied dev teamvendor with
support and SLA
 SIPDROID - only Android platform, no centralizied dev teamvendor with
support and SLA
 Counterpath - as i know no mobile platforms 

Searching...  

On 08/16/2012 08:58 PM, freeswitch wrote:   IMSDROID will be good!

  At 2012-08-16 20:44:54,BookBag  [4] wrote:
 I'm surprised no one said Bria by counterpath. Which allows for xmpp and
presence. The softphone allows you to send sms also if you buy the full
version. Another one that I tested is Csipsimple, this one allows sms for
free. But my absolute favorite is Zoiper, unfortunately no sms capability but
wasnt buggy except for one bug I encountered with sending the correct dtmf
style. Thank god their new update took care of that. Phone now seems very
solid and now includes a QR reader.

 On Thu, Aug 16, 2012 at 6:16 AM, vaad.fabi at gmail.com [5]  wrote:
  Does they provide these software development, customisation and
 support ? i think no. I need vendordevelopment team who can provide
 developmentcustomisationsupport.
 Anyway thx.

 On 08/16/2012 12:53 PM, Mohammad Emran wrote:
 > Then try nimbuzz or fring.... If it works for u.
 >
 > Sent from my iPhone
 >
 > On Aug 16, 2012, at 3:42 PM, "vaad.fabi at gmail.com [7]" wrote:
 >
 >> Seems to be ok for a simple dialer, but i forgot to mention - i need
 >> advanced options for dialers, such as: IM (xmppfacebooktwitter
 >> integration), video calls, balance state, check direction rate,
 >> callback, sms, top up and etc. It's smth like UC application with
 >> calling support as one of the many options.
 >>
 >>
 >> On 08/16/2012 12:22 PM, Mohammad Emran wrote:
 >>> Try www.sipmobiledialer.com [9]
 >>>
 >>>
 >>> Sent from my iPad
 >>>
 >>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com [10]" wrote:
 >>>
 >>>> Hi all,
 >>>>
 >>>> Please suggest mobile voip dialer vendor for
 >>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
 >>>> don't suggest google plz
 >>>>
 >>>> --
 >>>> Best Regards,
 >>>> Vadim F.
 >>>>
 >>>>
 >>>>
_________________________________________________________________________
 >>>> Professional FreeSWITCH Consulting Services:
 >>>> consulting at freeswitch.org [12]
 >>>> http://www.freeswitchsolutions.com [13]
 >>>>
 >>>> 
 >>>>  [14]
 >>>>
 >>>> Official FreeSWITCH Sites
 >>>> http://www.freeswitch.org [15]
 >>>> http://wiki.freeswitch.org [16]
 >>>> http://www.cluecon.com [17]
 >>>>
 >>>> FreeSWITCH-users mailing list
 >>>> FreeSWITCH-users at lists.freeswitch.org [18]
 >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [19]
 >>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [20]
 >>>> http://www.freeswitch.org [21]
 >>>
_________________________________________________________________________
 >>> Professional FreeSWITCH Consulting Services:
 >>> consulting at freeswitch.org [22]
 >>> http://www.freeswitchsolutions.com [23]
 >>>
 >>> 
 >>>  [24]
 >>>
 >>> Official FreeSWITCH Sites
 >>> http://www.freeswitch.org [25]
 >>> http://wiki.freeswitch.org [26]
 >>> http://www.cluecon.com [27]
 >>>
 >>> FreeSWITCH-users mailing list
 >>> FreeSWITCH-users at lists.freeswitch.org [28]
 >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [29]
 >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[30]
 >>> http://www.freeswitch.org [31]
 >>
 >> --
 >> Best Regards,
 >> Vadim F.
 >>
 >>
 >> _________________________________________________________________________
 >> Professional FreeSWITCH Consulting Services:
 >> consulting at freeswitch.org [32]
 >> http://www.freeswitchsolutions.com [33]
 >>
 >> 
 >>  [34]
 >>
 >> Official FreeSWITCH Sites
 >> http://www.freeswitch.org [35]
 >> http://wiki.freeswitch.org [36]
 >> http://www.cluecon.com [37]
 >>
 >> FreeSWITCH-users mailing list
 >> FreeSWITCH-users at lists.freeswitch.org [38]
 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [39]
 >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[40]
 >> http://www.freeswitch.org [41]
 > _________________________________________________________________________
 > Professional FreeSWITCH Consulting Services:
 > consulting at freeswitch.org [42]
 > http://www.freeswitchsolutions.com [43]
 >
 > 
 >  [44]
 >
 > Official FreeSWITCH Sites
 > http://www.freeswitch.org [45]
 > http://wiki.freeswitch.org [46]
 > http://www.cluecon.com [47]
 >
 > FreeSWITCH-users mailing list
 > FreeSWITCH-users at lists.freeswitch.org [48]
 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [49]
 > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[50]
 > http://www.freeswitch.org [51]

 --
 Best Regards,
 Vadim F.

 _________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 consulting at freeswitch.org [52]
 http://www.freeswitchsolutions.com [53]

 
  [54]

 Official FreeSWITCH Sites
 http://www.freeswitch.org [55]
 http://wiki.freeswitch.org [56]
 http://www.cluecon.com [57]

 FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org [58]
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [59]
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[60]
 http://www.freeswitch.org [61]      

 _________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting at freeswitch.org [62]
http://www.freeswitchsolutions.com [63] FreeSWITCH-powered IP PBX: The
CudaTel Communication Server  [64] Official FreeSWITCH
Sites http://www.freeswitch.org [65] http://wiki.freeswitch.org [66]
http://www.cluecon.com [67] FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org [68]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [69]
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [70]
http://www.freeswitch.org [71]   

 -- Best Regards, Vadim F.    
_________________________________________________________________________
 Professional FreeSWITCH Consulting Services:
 consulting at freeswitch.org [72]
 http://www.freeswitchsolutions.com [73]

 
  [74]

 Official FreeSWITCH Sites
 http://www.freeswitch.org [75]
 http://wiki.freeswitch.org [76]
 http://www.cluecon.com [77]

 FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org [78]
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [79]
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[80]
 http://www.freeswitch.org [81]

-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
 Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com [82]
Email: shaheryarkh at googlemail.com [83] 

Links:
------
[1] http://www.doubango.org
[2] mailto:vaad.fabi at gmail.com
[3] mailto:vaad.fabi at gmail.com
[4] mailto:asaad2 at gmail.com
[5] mailto:vaad.fabi at gmail.com
[6] mailto:vaad.fabi at gmail.com
[7] mailto:vaad.fabi at gmail.com
[8] mailto:vaad.fabi at gmail.com
[9] http://www.sipmobiledialer.com
[10] mailto:vaad.fabi at gmail.com
[11] mailto:vaad.fabi at gmail.com
[12] mailto:consulting at freeswitch.org
[13] http://www.freeswitchsolutions.com
[14] 
[15] http://www.freeswitch.org
[16] http://wiki.freeswitch.org
[17] http://www.cluecon.com
[18] mailto:FreeSWITCH-users at lists.freeswitch.org
[19] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[20] http://lists.freeswitch.org/mailman/options/freeswitch-users
[21] http://www.freeswitch.org
[22] mailto:consulting at freeswitch.org
[23] http://www.freeswitchsolutions.com
[24] 
[25] http://www.freeswitch.org
[26] http://wiki.freeswitch.org
[27] http://www.cluecon.com
[28] mailto:FreeSWITCH-users at lists.freeswitch.org
[29] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[30] http://lists.freeswitch.org/mailman/options/freeswitch-users
[31] http://www.freeswitch.org
[32] mailto:consulting at freeswitch.org
[33] http://www.freeswitchsolutions.com
[34] 
[35] http://www.freeswitch.org
[36] http://wiki.freeswitch.org
[37] http://www.cluecon.com
[38] mailto:FreeSWITCH-users at lists.freeswitch.org
[39] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[40] http://lists.freeswitch.org/mailman/options/freeswitch-users
[41] http://www.freeswitch.org
[42] mailto:consulting at freeswitch.org
[43] http://www.freeswitchsolutions.com
[44] 
[45] http://www.freeswitch.org
[46] http://wiki.freeswitch.org
[47] http://www.cluecon.com
[48] mailto:FreeSWITCH-users at lists.freeswitch.org
[49] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[50] http://lists.freeswitch.org/mailman/options/freeswitch-users
[51] http://www.freeswitch.org
[52] mailto:consulting at freeswitch.org
[53] http://www.freeswitchsolutions.com
[54] 
[55] http://www.freeswitch.org
[56] http://wiki.freeswitch.org
[57] http://www.cluecon.com
[58] mailto:FreeSWITCH-users at lists.freeswitch.org
[59] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[60] http://lists.freeswitch.org/mailman/options/freeswitch-users
[61] http://www.freeswitch.org
[62] mailto:consulting at freeswitch.org
[63] http://www.freeswitchsolutions.com
[64] 
[65] http://www.freeswitch.org
[66] http://wiki.freeswitch.org
[67] http://www.cluecon.com
[68] mailto:FreeSWITCH-users at lists.freeswitch.org
[69] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[70] http://lists.freeswitch.org/mailman/options/freeswitch-users
[71] http://www.freeswitch.org
[72] mailto:consulting at freeswitch.org
[73] http://www.freeswitchsolutions.com
[74] 
[75] http://www.freeswitch.org
[76] http://wiki.freeswitch.org
[77] http://www.cluecon.com
[78] mailto:FreeSWITCH-users at lists.freeswitch.org
[79] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[80] http://lists.freeswitch.org/mailman/options/freeswitch-users
[81] http://www.freeswitch.org
[82] mailto:shari_786pk at hotmail.com
[83] mailto:shaheryarkh at googlemail.com


____________________________________________________________
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From brett at launch3.net  Fri Aug 17 16:18:31 2012
From: brett at launch3.net (Brett Wilson)
Date: Fri, 17 Aug 2012 08:18:31 -0400
Subject: [Freeswitch-users] mod_portaudio cannot find input or output device
Message-ID: <021301cd7c72$6adff2f0$409fd8d0$@launch3.net>

Hey guys maybe you can help with this. Been trying to get portaudio working,
but cannot. I have built from the latest git source. Even downloaded and
build portaudio, which sees the device fine. Also, sound output works
through aplay on the command line. So why can't mod_portaudio find the
devices?

 

# lsmod | grep snd

snd_intel8x0           33455  0

snd_ac97_codec        106082  1 snd_intel8x0

ac97_bus               12642  1 snd_ac97_codec

snd_pcm                80845  2 snd_intel8x0,snd_ac97_codec

snd_seq_midi           13132  0

snd_rawmidi            25424  1 snd_seq_midi

snd_seq_midi_event     14475  1 snd_seq_midi

snd_seq                51567  2 snd_seq_midi,snd_seq_midi_event

snd_timer              28931  2 snd_pcm,snd_seq

snd_seq_device         14172  3 snd_seq_midi,snd_rawmidi,snd_seq

snd                    62064  7
snd_intel8x0,snd_ac97_codec,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_de
vice

soundcore              14635  1 snd

snd_page_alloc         14108  2 snd_intel8x0,snd_pcm

 

# lspci

00:00.0 Host bridge: Intel Corporation 82845G/GL[Brookdale-G]/GE/PE DRAM
Controller/Host-Hub Interface (rev 01)

00:02.0 VGA compatible controller: Intel Corporation
82845G/GL[Brookdale-G]/GE Chipset Integrated Graphics Device (rev 01)

00:1d.0 USB controller: Intel Corporation 82801DB/DBL/DBM
(ICH4/ICH4-L/ICH4-M) USB UHCI Controller #1 (rev 01)

00:1d.1 USB controller: Intel Corporation 82801DB/DBL/DBM
(ICH4/ICH4-L/ICH4-M) USB UHCI Controller #2 (rev 01)

00:1d.2 USB controller: Intel Corporation 82801DB/DBL/DBM
(ICH4/ICH4-L/ICH4-M) USB UHCI Controller #3 (rev 01)

00:1d.7 USB controller: Intel Corporation 82801DB/DBM (ICH4/ICH4-M) USB2
EHCI Controller (rev 01)

00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev 81)

00:1f.0 ISA bridge: Intel Corporation 82801DB/DBL (ICH4/ICH4-L) LPC
Interface Bridge (rev 01)

00:1f.1 IDE interface: Intel Corporation 82801DB (ICH4) IDE Controller (rev
01)

00:1f.3 SMBus: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) SMBus
Controller (rev 01)

00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM
(ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 01)

01:0c.0 Ethernet controller: Intel Corporation 82540EM Gigabit Ethernet
Controller (rev 02)

 

# cat /etc/group

audio:x:29:freeswitch,www-data

www-data:x:33:freeswitch

 

freeswitch at internal> load mod_portaudio

+OK Reloading XML

-ERR [module load file routine returned an error]

 

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1767 Cannot find an input
device

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1777 Cannot find an output
device

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1422 Loading streams ...

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1454 Parsing stream
'usb1' parameter indev = #2

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1458 Invalid indev
specified for stream 'usb1'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1454 Parsing stream
'usb1' parameter outdev = #2

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1467 Invalid outdev
specified for stream 'usb1'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1454 Parsing stream
'usb1' parameter sample-rate = 48000

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1454 Parsing stream
'usb1' parameter codec-ms = 10

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1454 Parsing stream
'usb1' parameter channels = 2

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1498 You need at least one
device for stream 'usb1'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1454 Parsing stream
'default' parameter indev = #0

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1458 Invalid indev
specified for stream 'default'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1454 Parsing stream
'default' parameter outdev = #1

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1467 Invalid outdev
specified for stream 'default'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1454 Parsing stream
'default' parameter sample-rate = 48000

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1454 Parsing stream
'default' parameter codec-ms = 10

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1454 Parsing stream
'default' parameter channels = 1

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1498 You need at least one
device for stream 'default'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1569 Loading endpoints
...

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1598 Parsing endpoint
'default' parameter instream = default:0

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1603 Invalid instream
specified for endpoint 'default'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1598 Parsing endpoint
'default' parameter outstream = default:0

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1611 Invalid outstream
specified for endpoint 'default'

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1620 You need at least one
stream for endpoint 'default'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1598 Parsing endpoint
'usb1out-left' parameter outstream = usb1:0

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1611 Invalid outstream
specified for endpoint 'usb1out-left'

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1620 You need at least one
stream for endpoint 'usb1out-left'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1598 Parsing endpoint
'usb1out-right' parameter outstream = usb1:1

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1611 Invalid outstream
specified for endpoint 'usb1out-right'

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1620 You need at least one
stream for endpoint 'usb1out-right'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1598 Parsing endpoint
'usb1in-left' parameter instream = usb1:0

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1603 Invalid instream
specified for endpoint 'usb1in-left'

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1620 You need at least one
stream for endpoint 'usb1in-left'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1598 Parsing endpoint
'usb1in-right' parameter instream = usb1:1

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1603 Invalid instream
specified for endpoint 'usb1in-right'

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1620 You need at least one
stream for endpoint 'usb1in-right'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1598 Parsing endpoint
'usb1-left' parameter instream = usb1:0

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1603 Invalid instream
specified for endpoint 'usb1-left'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1598 Parsing endpoint
'usb1-left' parameter outstream = usb1:0

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1611 Invalid outstream
specified for endpoint 'usb1-left'

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1620 You need at least one
stream for endpoint 'usb1-left'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1598 Parsing endpoint
'usb1-right' parameter instream = usb1:1

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1603 Invalid instream
specified for endpoint 'usb1-right'

2012-08-17 08:15:23.591770 [DEBUG] mod_portaudio.c:1598 Parsing endpoint
'usb1-right' parameter outstream = usb1:1

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1611 Invalid outstream
specified for endpoint 'usb1-right'

2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1620 You need at least one
stream for endpoint 'usb1-right'

2012-08-17 08:15:23.591770 [CRIT] switch_loadable_module.c:1310 Error
Loading module /usr/local/freeswitch/mod/mod_portaudio.so

**Module load routine returned an error**

******************************************* 
Brett Wilson
IT Department
Launch 3 Ventures, LLC 
134 Myer Street 
Hackensack, NJ 07601 
Phone:  <tel:877.878.9134> 877.878.9134
Fax:  <tel:646.536.3866> 646.536.3866 
Email:  <mailto:Brett.Wilson at launch3.net> Brett.Wilson at launch3.net
AOL IM:  <mailto:Brett.Wilson at launch3.net> Brett.Wilson at launch3.net
 <http://www.launch3.net/> www.Launch3.net 
 <http://www.launch3telecom.com/> www.Launch3telecom.com 
******************************************* 
 <http://launch3telecom.blogspot.com/> Description: Description:
Description: Description: Blogger-logo
<http://www.facebook.com/pages/Launch-3-Telecom/165610820178720>
Description: Description: Description: Description: FaceBook-Logo
<https://twitter.com/launch3telecom> Description: Description: Description:
Description: Twitter-Logo <https://plus.google.com/102713212642017290959>
Description: Description: Description: Description: GPlus-Logo

 

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From freeswitch-list at puzzled.xs4all.nl  Fri Aug 17 16:34:10 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 17 Aug 2012 14:34:10 +0200
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <CAMwZ-t+KUp494GT90dV7oS02sviMVerxJBNCVtyRosg_qneBRA@mail.gmail.com>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<1345176410.502dc35a7c3ec@mx.newgen.co.in>
	<CAMwZ-t+KUp494GT90dV7oS02sviMVerxJBNCVtyRosg_qneBRA@mail.gmail.com>
Message-ID: <502E3A42.5060201@puzzled.xs4all.nl>

On 17-08-12 08:58, Gabriel Gunderson wrote:
> On Thu, Aug 16, 2012 at 10:06 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:
>> Please help.
>
> You're going to need to be more patient than that. You let 17 full
> hours pass. You need to give people time to sleep and do their day
> jobs.

Or hire a consultant: consulting at freeswitch.org or +1-213-286-0400.

> Anyway, good luck.

Ditto.

Patrick



From freeswitch-list at puzzled.xs4all.nl  Fri Aug 17 16:45:27 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 17 Aug 2012 14:45:27 +0200
Subject: [Freeswitch-users] mod_portaudio cannot find input or output
 device
In-Reply-To: <021301cd7c72$6adff2f0$409fd8d0$@launch3.net>
References: <021301cd7c72$6adff2f0$409fd8d0$@launch3.net>
Message-ID: <502E3CE7.4070902@puzzled.xs4all.nl>

On 17-08-12 14:18, Brett Wilson wrote:
[snip]
> *freeswitch at internal> load mod_portaudio*
>
> +OK Reloading XML
>
> -ERR [module load file routine returned an error]

That ERR does not look good. Iirc I have seen that with mod_java when 
starting FreeSWITCH. Subsequent manual loading of the mod_java module 
went fine. But in your case you are already manually loading the module.

> *2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1767 Cannot find an
> input device*
>
> *2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1777 Cannot find an
> output device*

Did you set the input/output devices? Lot's of info on the wiki:

http://wiki.freeswitch.org/wiki/FreeSwitch_Enpoint_Portaudio#List_the_available_devices

Regards,
Patrick



From g.d.monnezza at tiscali.it  Fri Aug 17 16:56:23 2012
From: g.d.monnezza at tiscali.it (g)
Date: Fri, 17 Aug 2012 14:56:23 +0200
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
Message-ID: <10440306.PhsNeVk7eG@virtex>

Me too, I'm often dealing with callcenters (inbound+outbound). I found in 
Freeswitch the easyest solution to manage queue and agents. I'm not a 
programmer, but I have experience with Asterisk, Freeswitch and the whole 
workflow of complex callcenters.
My suggestion too is to use mod_callcenter. Easy, fast, powerful. You don't 
need agent to dial anything to get the call from queue.
In my opinion, you can use that module or, for more complex things (ie. 
outbound predictive) you can use the socket manager to perform  call 
placement, call transfer, conference with thirdy-part recording  IVR etc.
I feel the best solution, also if it will require some effort for a Java 
programmer, is to do all that in php, which is light and fast also for older  
PC, able to run also with thin-client low memory low-cpu machines.
If you can plan to turn to php and you don't have too short time, I can 
consider to help for the Freeswitch part.
giuliano


On Thursday 16 August 2012 17:10:33 Nitin Tomer wrote:
> Hi,
> 
> 
> 
> I am working on a Contact Center solution. It will support mail, chat and
> call queries.
> 
> The requirements are:
> 
> 1. An end-customer calls, the call is handled by FreeSWITCH Auto Attendant.
> 2. Customer is presented with a menu and makes selection. His call is put on
> hold and an entry is made in my system's database for incoming queries. 3.
> These queries are shown to agents handling calls.
> 4. And agent clicks on a query, he is shown an extension where call is
> parked. He dials that and is connected to the customer.
> 
> 5. He talks to the customer and resolve his queries.
> 
> Please guide me on how can do it. The application will be written in Java.
> 
> I am an experienced programmers in Java/J2EE; but doesn't have much
> knowledge of VoIP/FreeSWITCH. I've configured FreeSWITCH on Ubuntu 12.04;
> and have installed X-Lite Softphones on two Windows machines. I configured
> these phones to work with FreeSWITCH and they are working fine.
> 
> 
> 
> I think that I'd have to use Valet Parking to park customer's call on an
> extension, then pass that extension to agent who will dial it and will be
> connected to the customer.
> 
> 
> 
> Please tell me whether this approach will work? And how should I go about
> it.
> 
> 
> 
> All help would be much appreciated.
> 
> 
> 
> Regards
> 
> 
> 
> Nitin


From x.liu at hw.ac.uk  Fri Aug 17 17:48:04 2012
From: x.liu at hw.ac.uk (x.liu)
Date: Fri, 17 Aug 2012 14:48:04 +0100
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <10440306.PhsNeVk7eG@virtex>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<10440306.PhsNeVk7eG@virtex>
Message-ID: <502E4B94.1040503@hw.ac.uk>

I am also trying the mod_callcenter. What I want is to route a call to 
one agent ( a registered FS client, e.g. a SIP softphone)
of a list of agents by a Random or RoundRobin policy.

When I dial the extension which calls the callcenter app, I only have 
hold-on music. It does not route a call to a agent.
You said " You don't need agent to dial anything to get the call from 
queue. ",
I am wondering how I could let an agent log into the queue?

Thanks!

Xing


On 08/17/2012 01:56 PM, g wrote:
> Me too, I'm often dealing with callcenters (inbound+outbound). I found in
> Freeswitch the easyest solution to manage queue and agents. I'm not a
> programmer, but I have experience with Asterisk, Freeswitch and the whole
> workflow of complex callcenters.
> My suggestion too is to use mod_callcenter. Easy, fast, powerful. You don't
> need agent to dial anything to get the call from queue.
> In my opinion, you can use that module or, for more complex things (ie.
> outbound predictive) you can use the socket manager to perform  call
> placement, call transfer, conference with thirdy-part recording  IVR etc.
> I feel the best solution, also if it will require some effort for a Java
> programmer, is to do all that in php, which is light and fast also for older
> PC, able to run also with thin-client low memory low-cpu machines.
> If you can plan to turn to php and you don't have too short time, I can
> consider to help for the Freeswitch part.
> giuliano
>
>
> On Thursday 16 August 2012 17:10:33 Nitin Tomer wrote:
>> Hi,
>>
>>
>>
>> I am working on a Contact Center solution. It will support mail, chat and
>> call queries.
>>
>> The requirements are:
>>
>> 1. An end-customer calls, the call is handled by FreeSWITCH Auto Attendant.
>> 2. Customer is presented with a menu and makes selection. His call is put on
>> hold and an entry is made in my system's database for incoming queries. 3.
>> These queries are shown to agents handling calls.
>> 4. And agent clicks on a query, he is shown an extension where call is
>> parked. He dials that and is connected to the customer.
>>
>> 5. He talks to the customer and resolve his queries.
>>
>> Please guide me on how can do it. The application will be written in Java.
>>
>> I am an experienced programmers in Java/J2EE; but doesn't have much
>> knowledge of VoIP/FreeSWITCH. I've configured FreeSWITCH on Ubuntu 12.04;
>> and have installed X-Lite Softphones on two Windows machines. I configured
>> these phones to work with FreeSWITCH and they are working fine.
>>
>>
>>
>> I think that I'd have to use Valet Parking to park customer's call on an
>> extension, then pass that extension to agent who will dial it and will be
>> connected to the customer.
>>
>>
>>
>> Please tell me whether this approach will work? And how should I go about
>> it.
>>
>>
>>
>> All help would be much appreciated.
>>
>>
>>
>> Regards
>>
>>
>>
>> Nitin
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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



-- 
Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012

We invite research leaders and ambitious early career researchers to 
join us in leading and driving research in key inter-disciplinary themes. 
Please see www.hw.ac.uk/researchleaders for further information and how
to apply.

Heriot-Watt University is a Scottish charity
registered under charity number SC000278.



From mitch.capper at gmail.com  Fri Aug 17 18:48:29 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Fri, 17 Aug 2012 07:48:29 -0700
Subject: [Freeswitch-users] mod_portaudio cannot find input or output
	device
In-Reply-To: <502E3CE7.4070902@puzzled.xs4all.nl>
References: <021301cd7c72$6adff2f0$409fd8d0$@launch3.net>
	<502E3CE7.4070902@puzzled.xs4all.nl>
Message-ID: <CALn233qPLVkk3efB+VPeWZgw_0ziqOWTOo137oTidnPFJg1UsQ@mail.gmail.com>

The first problem may be the default config you are using is going to
cause the console to load with a lot of errors that may just add
confusion.  Portaudio streams are enabled but not documented by
default and are throwing atleast some of those errors (and for most
are not needed).   Try a much simple PA config like:
                <configuration name="portaudio.conf"
description="Soundcard Endpoint">
                  <settings>
                    <param name="ring-file"
value="tone_stream://%(2000,4000,440.0,480.0);loops=20"/>
                    <param name="ring-interval" value="5"/>
                    <param name="hold-file" value="$${hold_music}"/>
                    <param name="dialplan" value="XML"/>
                    <param name="sample-rate" value="24000"/>
                    <param name="codec-ms" value="20"/>
                  </settings>
                </configuration>

Then see if you can get portaudio loading, however I believe the
inability to find any audio devices is kicking the error out.  You
would want to try running aplay -l from a command line to make sure
some audio devices are listed.  In addition try running freeswitch as
root without the -u flag and ensure that its not a permissions
problem.

~Mitch

On Fri, Aug 17, 2012 at 5:45 AM, Patrick Lists
<freeswitch-list at puzzled.xs4all.nl> wrote:
> On 17-08-12 14:18, Brett Wilson wrote:
> [snip]
>> *freeswitch at internal> load mod_portaudio*
>>
>> +OK Reloading XML
>>
>> -ERR [module load file routine returned an error]
>
> That ERR does not look good. Iirc I have seen that with mod_java when
> starting FreeSWITCH. Subsequent manual loading of the mod_java module
> went fine. But in your case you are already manually loading the module.
>
>> *2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1767 Cannot find an
>> input device*
>>
>> *2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1777 Cannot find an
>> output device*
>
> Did you set the input/output devices? Lot's of info on the wiki:
>
> http://wiki.freeswitch.org/wiki/FreeSwitch_Enpoint_Portaudio#List_the_available_devices
>
> Regards,
> Patrick
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 onweike at esoftiesnigeria.com  Fri Aug 17 14:22:40 2012
From: onweike at esoftiesnigeria.com (Nweike Onwuyali)
Date: Fri, 17 Aug 2012 11:22:40 +0100
Subject: [Freeswitch-users] Aliased State Not showing in my sofia status
Message-ID: <001401cd7c62$3c1f35d0$b45da170$@com>

Hi All,

I am a newbie to freeswitch and I have downloaded the windows edition and
have it running on my XP Pro. However I have also downloaded a free DNS dual
server and installed as well. I have not been able to register the Yealink
T22P phone or snom 320 phone. 

I noticed that when I ran the sofia status command  I did not see any alias
type.

 

What should I do?

 

Cheers

Nweike 

 

 

 

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From lists at telefaks.de  Fri Aug 17 19:03:13 2012
From: lists at telefaks.de (Peter Steinbach)
Date: Fri, 17 Aug 2012 17:03:13 +0200
Subject: [Freeswitch-users] displace_session and moh via event socket
Message-ID: <502E5D31.9010203@telefaks.de>

Hello,

In an IVR we want to play moh for 10 seconds via event socket to an
existing session. We send the following message (we actually do not use
ESL for certain reasons):
  SendMsg
  call-command: execute
  execute-app-name: displace_session
  execute-app-arg: local_stream://moh m +10000
  event-lock:false

I think, this is the only way to play a stream for a certain time within
a session right?

We get the following output:
2012-08-17 16:05:20.261460 [DEBUG] switch_ivr.c:599
sofia/internal/202 at fs100.mydomain.de Command Execute
displace_session(local_stream://moh m +10000)
EXECUTE sofia/internal/202 at fs100.mydomain.de
displace_session(local_stream://moh m +10000)
2012-08-17 16:05:20.261460 [DEBUG] mod_local_stream.c:417 Opening Stream
[moh/8000] 8000hz
2012-08-17 16:05:20.261460 [DEBUG] switch_core_media_bug.c:502 Attaching
BUG to sofia/internal/202 at fs100.mydomain.de
2012-08-17 16:05:20.321460 [DEBUG] switch_core_io.c:353 Setting BUG
Codec PCMA:8
2012-08-17 16:05:27.981463 [CRIT] mod_local_stream.c:297 Leaking stream
handle! [switch_ivr_displace_session() src/switch_ivr_async.c:951]
2012-08-17 16:05:35.701462 [CRIT] mod_local_stream.c:297 Leaking stream
handle! [switch_ivr_displace_session() src/switch_ivr_async.c:951]
2012-08-17 16:05:43.421463 [CRIT] mod_local_stream.c:297 Leaking stream
handle! [switch_ivr_displace_session() src/switch_ivr_async.c:951]
and no sound is played.

Passing a regular filename does not work either. Without the "m" or time
parameter we have also the same result: No file is played. See the
following log:
2012-08-17 17:00:51.911469 [DEBUG] switch_ivr.c:599
sofia/internal/202 at fs100.mydomain.de Command Execute
displace_session(/usr/local/freeswitch/sounds/en/us/callie/google/de/willkommen.mp3
m +10000)
EXECUTE sofia/internal/202 at fs100.mydomain.de
displace_session(/usr/local/freeswitch/sounds/en/us/callie/google/de/willkommen.mp3
m +10000)
2012-08-17 17:00:51.921461 [DEBUG] switch_core_media_bug.c:502 Attaching
BUG to sofia/internal/202 at fs100.mydomain.de
2012-08-17 17:00:51.941460 [DEBUG] switch_core_io.c:353 Setting BUG
Codec PCMA:8
And then output stops for this session.

An normal playback works however, but with playback we cannot limit a
file to a certain time period, right?

Did Anybody have the same issue and knows how to solve this?


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From krice at freeswitch.org  Fri Aug 17 19:18:21 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 17 Aug 2012 10:18:21 -0500
Subject: [Freeswitch-users] Weekly Conf Call Reminder
Message-ID: <CC53CAED.3C13A%krice@freeswitch.org>

Hey Guys,

We?re always looking for new ideas and guest speakers to have on the Weekly
FreeSWITCH Community Conference Call. If you have any ideas or requests, Let
us Know!

How can you let us know this?
1. Contact me Offlist
2. Reply here with those ideas
3. update the wiki and add some ideas to the Suggestions for Future Meetings
sections.

What kind of things are appropriate? Well pretty much anything related to
FreeSWITCH. Some examples of previous discussions:
SIP101
Vestec ASR (Speech Rec engine)
Giovanni Maruzzelli presenting on GSMOpen
Travis Cross discussing GIT
Goni AKA Mahammed Naseer presenting and discussing vBilling
Mr Phil Zimmerman discussing ZRTP and VoIP Security.


As you will notice not all of these subjects like GIT are about FreeSWITCH,
but each and everyone include GIT is relevant to FreeSWITCH.

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From msc at freeswitch.org  Fri Aug 17 19:18:31 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 17 Aug 2012 08:18:31 -0700
Subject: [Freeswitch-users] Aliased State Not showing in my sofia status
In-Reply-To: <001401cd7c62$3c1f35d0$b45da170$@com>
References: <001401cd7c62$3c1f35d0$b45da170$@com>
Message-ID: <CAKzWOxXhnxSjPphJ9Zm-KeJFFWBEtt6QK2wkix0xRTLbB9OwVA@mail.gmail.com>

Paste us the output from "sofia status"
-MC

On Fri, Aug 17, 2012 at 3:22 AM, Nweike Onwuyali <
onweike at esoftiesnigeria.com> wrote:

> Hi All,****
>
> I am a newbie to freeswitch and I have downloaded the windows edition and
> have it running on my XP Pro. However I have also downloaded a free DNS
> dual server and installed as well. I have not been able to register the
> Yealink  T22P phone or snom 320 phone. ****
>
> I noticed that when I ran the sofia status command  I did not see any
> alias type.****
>
> ** **
>
> What should I do?****
>
> ** **
>
> Cheers****
>
> Nweike ****
>
> ** **
>
> ** **
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From ben at langfeld.co.uk  Fri Aug 17 19:38:04 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Fri, 17 Aug 2012 16:38:04 +0100
Subject: [Freeswitch-users] Weekly Conf Call Reminder
In-Reply-To: <CC53CAED.3C13A%krice@freeswitch.org>
References: <CC53CAED.3C13A%krice@freeswitch.org>
Message-ID: <CAAyX+KHuCCj5W_itAPbfx0AEk+52L7+1xr6Sj30JzOCpTt8vTw@mail.gmail.com>

This year we released Adhearsion 2.1 at ClueCon where we added support for
FreeSWITCH. Adhearsion now boasts support for 3 VoIP engines (FreeSWITCH,
Asterisk and Voxeo PRISM), allowing a degree of portability between them.
We hope this will encourage some people who currently use Asterisk with
Adhearsion to give FreeSWITCH a try instead.

For those who don't know, Adhearsion is a framework for developing voice
applications, and it's written in Ruby. It connects over the network to one
of the above mentioned engines and exercises third-party control over
calls. It is great for writing voice applications integrated with other
services such as databases, web APIs, message queues, etc. It is used for
IVRs, Call Centre automation, data gathering tasks as well as enabling a
whole slew of "Voice 2.0" type applications.

To learn more, check out http://adhearsion.com, or fire me an email. Does
anyone want to hear about Adhearsion on the conference call?

Regards,
Ben Langfeld


On 17 August 2012 16:18, Ken Rice <krice at freeswitch.org> wrote:

>  Hey Guys,
>
> We?re always looking for new ideas and guest speakers to have on the
> Weekly FreeSWITCH Community Conference Call. If you have any ideas or
> requests, Let us Know!
>
> How can you let us know this?
>
>    1. Contact me Offlist
>    2. Reply here with those ideas
>    3. update the wiki and add some ideas to the Suggestions for Future
>    Meetings sections.
>
>
> What kind of things are appropriate? Well pretty much anything related to
> FreeSWITCH. Some examples of previous discussions:
> SIP101
> Vestec ASR (Speech Rec engine)
> Giovanni Maruzzelli presenting on GSMOpen
> Travis Cross discussing GIT
> Goni AKA Mahammed Naseer presenting and discussing vBilling
> Mr Phil Zimmerman discussing ZRTP and VoIP Security.
>
>
> As you will notice not all of these subjects like GIT are about
> FreeSWITCH, but each and everyone include GIT is relevant to FreeSWITCH.
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 krice at freeswitch.org  Fri Aug 17 19:55:01 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 17 Aug 2012 10:55:01 -0500
Subject: [Freeswitch-users] Weekly Conf Call Reminder
In-Reply-To: <CAAyX+KHuCCj5W_itAPbfx0AEk+52L7+1xr6Sj30JzOCpTt8vTw@mail.gmail.com>
Message-ID: <CC53D385.3C141%krice@freeswitch.org>

I was just speaking with bklang on IRC. Let us know when you guys can join
us on the call!


On 8/17/12 10:38 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:

> This year we released Adhearsion 2.1 at ClueCon where we added support for
> FreeSWITCH. Adhearsion now boasts support for 3 VoIP engines (FreeSWITCH,
> Asterisk and Voxeo PRISM), allowing a degree of portability between them. We
> hope this will encourage some people who currently use Asterisk with
> Adhearsion to give FreeSWITCH a try instead.
> 
> For those who don't know, Adhearsion is a framework for developing voice
> applications, and it's written in Ruby. It connects over the network to one of
> the above mentioned engines and exercises third-party control over calls. It
> is great for writing voice applications integrated with other services such as
> databases, web APIs, message queues, etc. It is used for IVRs, Call Centre
> automation, data gathering tasks as well as enabling a whole slew of "Voice
> 2.0" type applications.
> 
> To learn more, check out http://adhearsion.com, or fire me an email. Does
> anyone want to hear about Adhearsion on the conference call?
> 
> Regards,
> Ben Langfeld
> 
> 
> On 17 August 2012 16:18, Ken Rice <krice at freeswitch.org> wrote:
>> Hey Guys,
>> 
>> We?re always looking for new ideas and guest speakers to have on the Weekly
>> FreeSWITCH Community Conference Call. If you have any ideas or requests, Let
>> us Know!
>> 
>> How can you let us know this?
>> 1. Contact me Offlist
>> 2. Reply here with those ideas
>> 3. update the wiki and add some ideas to the Suggestions for Future Meetings
>> sections.
>> 
>> What kind of things are appropriate? Well pretty much anything related to
>> FreeSWITCH. Some examples of previous discussions:
>> SIP101
>> Vestec ASR (Speech Rec engine)
>> Giovanni Maruzzelli presenting on GSMOpen
>> Travis Cross discussing GIT
>> Goni AKA Mahammed Naseer presenting and discussing vBilling
>> Mr Phil Zimmerman discussing ZRTP and VoIP Security.
>> 
>> 
>> As you will notice not all of these subjects like GIT are about FreeSWITCH,
>> but each and everyone include GIT is relevant to FreeSWITCH.

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From jmesquita at freeswitch.org  Fri Aug 17 20:10:29 2012
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 17 Aug 2012 13:10:29 -0300
Subject: [Freeswitch-users] Weekly Conf Call Reminder
In-Reply-To: <CC53D385.3C141%krice@freeswitch.org>
References: <CAAyX+KHuCCj5W_itAPbfx0AEk+52L7+1xr6Sj30JzOCpTt8vTw@mail.gmail.com>
	<CC53D385.3C141%krice@freeswitch.org>
Message-ID: <CABH8QX1RFh=VfFW_cOvn3vZ5zCnsXODwN21UrjdoJ20CSgj47g@mail.gmail.com>

SwK, how about a more deep level discussion about FreeSWITCH internals? How
does the machine state works, which state does what, what are the flags for
the channels for and so on?

This is something that I believe most of us half-developers (no such term,
but you get the idea), would like to know about. It is quite challenging
sometimes to implement cool things with ESL without a knowledge of the
inner workings of FreeSWITCH and I can see that concepts such as loopback
and emulated att_xfer can easily be misleading if someone does not
understand those.

Anyhow, just an idea and I don't know how interested the core devs are in
doing this.

Regards,
Jo?o Mesquita



On Fri, Aug 17, 2012 at 12:55 PM, Ken Rice <krice at freeswitch.org> wrote:

>  I was just speaking with bklang on IRC. Let us know when you guys can
> join us on the call!
>
>
>
> On 8/17/12 10:38 AM, "Ben Langfeld" <ben at langfeld.co.uk> wrote:
>
> This year we released Adhearsion 2.1 at ClueCon where we added support for
> FreeSWITCH. Adhearsion now boasts support for 3 VoIP engines (FreeSWITCH,
> Asterisk and Voxeo PRISM), allowing a degree of portability between them.
> We hope this will encourage some people who currently use Asterisk with
> Adhearsion to give FreeSWITCH a try instead.
>
> For those who don't know, Adhearsion is a framework for developing voice
> applications, and it's written in Ruby. It connects over the network to one
> of the above mentioned engines and exercises third-party control over
> calls. It is great for writing voice applications integrated with other
> services such as databases, web APIs, message queues, etc. It is used for
> IVRs, Call Centre automation, data gathering tasks as well as enabling a
> whole slew of "Voice 2.0" type applications.
>
> To learn more, check out http://adhearsion.com, or fire me an email. Does
> anyone want to hear about Adhearsion on the conference call?
>
> Regards,
> Ben Langfeld
>
>
> On 17 August 2012 16:18, Ken Rice <krice at freeswitch.org> wrote:
>
> Hey Guys,
>
> We?re always looking for new ideas and guest speakers to have on the
> Weekly FreeSWITCH Community Conference Call. If you have any ideas or
> requests, Let us Know!
>
> How can you let us know this?
>
>    1. Contact me Offlist
>    2. Reply here with those ideas
>    3. update the wiki and add some ideas to the Suggestions for Future
>    Meetings sections.
>
>
> What kind of things are appropriate? Well pretty much anything related to
> FreeSWITCH. Some examples of previous discussions:
> SIP101
> Vestec ASR (Speech Rec engine)
> Giovanni Maruzzelli presenting on GSMOpen
> Travis Cross discussing GIT
> Goni AKA Mahammed Naseer presenting and discussing vBilling
> Mr Phil Zimmerman discussing ZRTP and VoIP Security.
>
>
> As you will notice not all of these subjects like GIT are about
> FreeSWITCH, but each and everyone include GIT is relevant to FreeSWITCH.
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 Hector.Geraldino at ipsoft.com  Fri Aug 17 20:30:04 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Fri, 17 Aug 2012 12:30:04 -0400
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <502E4B94.1040503@hw.ac.uk>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<10440306.PhsNeVk7eG@virtex> <502E4B94.1040503@hw.ac.uk>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD073ECD1635@NY1-EXMB-01.ip-soft.net>

In case you want to build your own queue instead of using mod_callcenter, you can:

1) playing a menu using the ivr application
2) after user selects an option, transfer the call to an extension that will send the control of the call to the java application (ESL in outbound socket mode)
3) java app inserts a record on the DB (with the call uuid), and parks the call (no valet parking, just park() )
4) your end-user app keeps polling the database, and shows the calls on hold
5) user clicks on the call record. This action sends a command to the java app that has control of the call (via RMI, sockets, HTTP, JMS, any communication method you choose). The message contains the call UUID and the extension of the user who clicked on the call record.
6) the java app receives the command, and performs a bridge between the call it is controlling and the extension received on the method call.

All this is completely doable. I myself have built an application to control service desk hotlines using FreeSWITCH + Java ESL client library 

http://wiki.freeswitch.org/wiki/Java_ESL_Client

g.luck!

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of x.liu
Sent: Friday, August 17, 2012 9:48 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Help needed regaridng Contact Center solution

I am also trying the mod_callcenter. What I want is to route a call to one agent ( a registered FS client, e.g. a SIP softphone) of a list of agents by a Random or RoundRobin policy.

When I dial the extension which calls the callcenter app, I only have hold-on music. It does not route a call to a agent.
You said " You don't need agent to dial anything to get the call from queue. ", I am wondering how I could let an agent log into the queue?

Thanks!

Xing


On 08/17/2012 01:56 PM, g wrote:
> Me too, I'm often dealing with callcenters (inbound+outbound). I found 
> in Freeswitch the easyest solution to manage queue and agents. I'm not 
> a programmer, but I have experience with Asterisk, Freeswitch and the 
> whole workflow of complex callcenters.
> My suggestion too is to use mod_callcenter. Easy, fast, powerful. You 
> don't need agent to dial anything to get the call from queue.
> In my opinion, you can use that module or, for more complex things (ie.
> outbound predictive) you can use the socket manager to perform  call 
> placement, call transfer, conference with thirdy-part recording  IVR etc.
> I feel the best solution, also if it will require some effort for a 
> Java programmer, is to do all that in php, which is light and fast 
> also for older PC, able to run also with thin-client low memory low-cpu machines.
> If you can plan to turn to php and you don't have too short time, I 
> can consider to help for the Freeswitch part.
> giuliano
>
>
> On Thursday 16 August 2012 17:10:33 Nitin Tomer wrote:
>> Hi,
>>
>>
>>
>> I am working on a Contact Center solution. It will support mail, chat 
>> and call queries.
>>
>> The requirements are:
>>
>> 1. An end-customer calls, the call is handled by FreeSWITCH Auto Attendant.
>> 2. Customer is presented with a menu and makes selection. His call is 
>> put on hold and an entry is made in my system's database for incoming queries. 3.
>> These queries are shown to agents handling calls.
>> 4. And agent clicks on a query, he is shown an extension where call 
>> is parked. He dials that and is connected to the customer.
>>
>> 5. He talks to the customer and resolve his queries.
>>
>> Please guide me on how can do it. The application will be written in Java.
>>
>> I am an experienced programmers in Java/J2EE; but doesn't have much 
>> knowledge of VoIP/FreeSWITCH. I've configured FreeSWITCH on Ubuntu 
>> 12.04; and have installed X-Lite Softphones on two Windows machines. 
>> I configured these phones to work with FreeSWITCH and they are working fine.
>>
>>
>>
>> I think that I'd have to use Valet Parking to park customer's call on 
>> an extension, then pass that extension to agent who will dial it and 
>> will be connected to the customer.
>>
>>
>>
>> Please tell me whether this approach will work? And how should I go 
>> about it.
>>
>>
>>
>> All help would be much appreciated.
>>
>>
>>
>> Regards
>>
>>
>>
>> Nitin
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-use
> rs
> http://www.freeswitch.org



--
Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012

We invite research leaders and ambitious early career researchers to join us in leading and driving research in key inter-disciplinary themes. 
Please see www.hw.ac.uk/researchleaders for further information and how to apply.

Heriot-Watt University is a Scottish charity registered under charity number SC000278.


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 vbvbrj at gmail.com  Fri Aug 17 20:32:01 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Fri, 17 Aug 2012 19:32:01 +0300
Subject: [Freeswitch-users] Recording problem
Message-ID: <502E7201.5050909@gmail.com>

Hello.

I have a problem with recorded files (wav, mp3, gsm) there are three 
agents in callcenter which receives calls from same external provider. 
The problem is that recorded sound from one agent is good quality, while 
other two recorded sounds have dropped samples and its like the sound is 
at 2x speed.

In vars.xml I have:
	<X-PRE-PROCESS cmd="set" 
data="global_codec_prefs=speex at 32000h@20i,G722,PCMU,PCMA,GSM"/>
	<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>

I am using mod_callcenter. There are 3 agents:
user 892 at D-Link DPH-150S
user 743 at D-Link DPH-150S
user 897 at Linksys PAP2T

There is incoming calls from same external number (multiple channels), 
the log is:
.....
a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.658288 [DEBUG] 
sofia.c:6060 Remote SDP:
a5079181-e072-4eee-9291-95a4ab40eabd v=0
a5079181-e072-4eee-9291-95a4ab40eabd o=- 1345205895 1345205895 IN IP4 
192.168.0.6
a5079181-e072-4eee-9291-95a4ab40eabd s=-
a5079181-e072-4eee-9291-95a4ab40eabd c=IN IP4 192.168.0.6
a5079181-e072-4eee-9291-95a4ab40eabd t=0 0
a5079181-e072-4eee-9291-95a4ab40eabd m=audio 13630 RTP/AVP 8 0
a5079181-e072-4eee-9291-95a4ab40eabd a=rtpmap:8 PCMA/8000
a5079181-e072-4eee-9291-95a4ab40eabd a=rtpmap:0 PCMU/8000
a5079181-e072-4eee-9291-95a4ab40eabd a=silenceSupp:off - - - -
a5079181-e072-4eee-9291-95a4ab40eabd
.....
a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[PCMA:8:8000:20:64000]/[SPEEX:99:32000:20:44000]
a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[PCMA:8:8000:20:64000]/[G722:9:8000:20:64000]
a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] 
sofia_glue.c:3027 Set Codec sofia/orange/extern_number at 192.168.0.5:5061 
PCMA/8000 20 ms 160 samples 64000 bits
a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] 
switch_core_codec.c:111 sofia/orange/extern_number at 192.168.0.5:5061 
Original read codec set to PCMA:8

After playing greeting, caller is routed to callcenter. When an agent 
answers the log has:

For user 892:
55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] 
sofia.c:6060 Remote SDP:
55969678-ab9f-4007-8023-304f4b602c97 v=0
55969678-ab9f-4007-8023-304f4b602c97 o=710 38421252 24417309 IN IP4 
132.101.208.92
55969678-ab9f-4007-8023-304f4b602c97 s=A conversation
55969678-ab9f-4007-8023-304f4b602c97 c=IN IP4 132.101.208.92
55969678-ab9f-4007-8023-304f4b602c97 t=0 0
55969678-ab9f-4007-8023-304f4b602c97 m=audio 10154 RTP/AVP 9 0 8 101
55969678-ab9f-4007-8023-304f4b602c97 a=rtpmap:9 G722/16000
55969678-ab9f-4007-8023-304f4b602c97 a=rtpmap:0 PCMU/8000
55969678-ab9f-4007-8023-304f4b602c97 a=rtpmap:8 PCMA/8000
55969678-ab9f-4007-8023-304f4b602c97 a=rtpmap:101 telephone-event/8000
55969678-ab9f-4007-8023-304f4b602c97 a=fmtp:101 0-15
......
55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[G722:9:16000:20:64000]/[SPEEX:99:32000:20:44000]
55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[G722:9:16000:20:64000]/[G722:9:8000:20:64000]
55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[G722:9:16000:20:64000]/[PCMU:0:8000:20:64000]
55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[G722:9:16000:20:64000]/[PCMA:8:8000:20:64000]
55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[G722:9:16000:20:64000]/[GSM:3:8000:20:13200]
55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] 
sofia_glue.c:5097 Substituting codec G722 at 20i@8000h
55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] 
sofia_glue.c:3027 Set Codec sofia/internal/sip:892 at 132.101.208.92:5060 
G722/8000 20 ms 160 samples 64000 bits
55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] 
switch_core_codec.c:111 sofia/internal/sip:892 at 132.101.208.92:5060 
Original read codec set to G722:9


For user 743:
acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] 
sofia.c:6060 Remote SDP:
acb5bf6a-4e79-46a4-a32b-420b5301d6ec v=0
acb5bf6a-4e79-46a4-a32b-420b5301d6ec o=1743 26961311 21406313 IN IP4 
132.101.207.43
acb5bf6a-4e79-46a4-a32b-420b5301d6ec s=A conversation
acb5bf6a-4e79-46a4-a32b-420b5301d6ec c=IN IP4 132.101.207.43
acb5bf6a-4e79-46a4-a32b-420b5301d6ec t=0 0
acb5bf6a-4e79-46a4-a32b-420b5301d6ec m=audio 10092 RTP/AVP 0 8 101
acb5bf6a-4e79-46a4-a32b-420b5301d6ec a=rtpmap:0 PCMU/8000
acb5bf6a-4e79-46a4-a32b-420b5301d6ec a=rtpmap:8 PCMA/8000
acb5bf6a-4e79-46a4-a32b-420b5301d6ec a=rtpmap:101 telephone-event/8000
acb5bf6a-4e79-46a4-a32b-420b5301d6ec a=fmtp:101 0-15
......
acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[PCMU:0:8000:20:64000]/[SPEEX:99:32000:20:44000]
acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] 
sofia_glue.c:3027 Set Codec sofia/internal/sip:743 at 132.101.207.43:5060 
PCMU/8000 20 ms 160 samples 64000 bits
acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] 
switch_core_codec.c:111 sofia/internal/sip:743 at 132.101.207.43:5060 
Original read codec set to PCMU:0


For user 897:
35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] 
sofia.c:6060 Remote SDP:
35c3b541-c78e-4d4d-96fe-f15509dc7f91 v=0
35c3b541-c78e-4d4d-96fe-f15509dc7f91 o=- 1715257 1715257 IN IP4 
132.101.202.2
35c3b541-c78e-4d4d-96fe-f15509dc7f91 s=-
35c3b541-c78e-4d4d-96fe-f15509dc7f91 c=IN IP4 132.101.202.2
35c3b541-c78e-4d4d-96fe-f15509dc7f91 t=0 0
35c3b541-c78e-4d4d-96fe-f15509dc7f91 m=audio 16444 RTP/AVP 0 100 101
35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=rtpmap:0 PCMU/8000
35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=rtpmap:100 NSE/8000
35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=fmtp:100 192-193
35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=rtpmap:101 telephone-event/8000
35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=fmtp:101 0-15
35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=ptime:30
......
35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[PCMU:0:8000:30:64000]/[SPEEX:99:32000:20:44000]
35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[PCMU:0:8000:30:64000]/[G722:9:8000:20:64000]
35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[PCMU:0:8000:30:64000]/[PCMU:0:8000:20:64000]
35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[PCMU:0:8000:30:64000]/[PCMA:8:8000:20:64000]
35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] 
sofia_glue.c:5044 Audio Codec Compare 
[PCMU:0:8000:30:64000]/[GSM:3:8000:20:13200]
35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] 
sofia_glue.c:5097 Substituting codec PCMU at 30i@8000h
35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] 
sofia_glue.c:3027 Set Codec sofia/internal/sip:897 at 132.101.202.2:5060 
PCMU/8000 30 ms 240 samples 64000 bits
35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] 
switch_core_codec.c:111 sofia/internal/sip:897 at 132.101.202.2:5060 
Original read codec set to PCMU:0



See the difference for users 892 and 743. For user 892 a substitution 
for codecs is done, while for user 743 is not. Both users are registered 
from the same model of IP phone with same configs except sip 
configuration for login.

What can be wrong?
Thank you.


From Hector.Geraldino at ipsoft.com  Fri Aug 17 20:34:49 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Fri, 17 Aug 2012 12:34:49 -0400
Subject: [Freeswitch-users] displace_session and moh via event socket
In-Reply-To: <502E5D31.9010203@telefaks.de>
References: <502E5D31.9010203@telefaks.de>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD073ECD1637@NY1-EXMB-01.ip-soft.net>

Have you tried a combination of playback + sched_api +10 uuid break ?

IIRC you can use break app to cancel the playback, so all you need is to schedule it to happen sometime in the future (10 seconds in your case)

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Steinbach
Sent: Friday, August 17, 2012 11:03 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] displace_session and moh via event socket

Hello,

In an IVR we want to play moh for 10 seconds via event socket to an existing session. We send the following message (we actually do not use ESL for certain reasons):
  SendMsg
  call-command: execute
  execute-app-name: displace_session
  execute-app-arg: local_stream://moh m +10000
  event-lock:false

I think, this is the only way to play a stream for a certain time within a session right?

We get the following output:
2012-08-17 16:05:20.261460 [DEBUG] switch_ivr.c:599 sofia/internal/202 at fs100.mydomain.de Command Execute displace_session(local_stream://moh m +10000) EXECUTE sofia/internal/202 at fs100.mydomain.de
displace_session(local_stream://moh m +10000)
2012-08-17 16:05:20.261460 [DEBUG] mod_local_stream.c:417 Opening Stream [moh/8000] 8000hz
2012-08-17 16:05:20.261460 [DEBUG] switch_core_media_bug.c:502 Attaching BUG to sofia/internal/202 at fs100.mydomain.de
2012-08-17 16:05:20.321460 [DEBUG] switch_core_io.c:353 Setting BUG Codec PCMA:8
2012-08-17 16:05:27.981463 [CRIT] mod_local_stream.c:297 Leaking stream handle! [switch_ivr_displace_session() src/switch_ivr_async.c:951]
2012-08-17 16:05:35.701462 [CRIT] mod_local_stream.c:297 Leaking stream handle! [switch_ivr_displace_session() src/switch_ivr_async.c:951]
2012-08-17 16:05:43.421463 [CRIT] mod_local_stream.c:297 Leaking stream handle! [switch_ivr_displace_session() src/switch_ivr_async.c:951] and no sound is played.

Passing a regular filename does not work either. Without the "m" or time parameter we have also the same result: No file is played. See the following log:
2012-08-17 17:00:51.911469 [DEBUG] switch_ivr.c:599 sofia/internal/202 at fs100.mydomain.de Command Execute
displace_session(/usr/local/freeswitch/sounds/en/us/callie/google/de/willkommen.mp3
m +10000)
EXECUTE sofia/internal/202 at fs100.mydomain.de
displace_session(/usr/local/freeswitch/sounds/en/us/callie/google/de/willkommen.mp3
m +10000)
2012-08-17 17:00:51.921461 [DEBUG] switch_core_media_bug.c:502 Attaching BUG to sofia/internal/202 at fs100.mydomain.de
2012-08-17 17:00:51.941460 [DEBUG] switch_core_io.c:353 Setting BUG Codec PCMA:8 And then output stops for this session.

An normal playback works however, but with playback we cannot limit a file to a certain time period, right?

Did Anybody have the same issue and knows how to solve this?


--
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 vbvbrj at gmail.com  Fri Aug 17 20:42:52 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Fri, 17 Aug 2012 19:42:52 +0300
Subject: [Freeswitch-users] Recording problem
In-Reply-To: <502E7201.5050909@gmail.com>
References: <502E7201.5050909@gmail.com>
Message-ID: <502E748C.7000900@gmail.com>

Forgot to mention. The user 743 has the good record audio file, the 
other two are bad recorded audio files.


From dgarcia at anew.com.ve  Fri Aug 17 21:05:17 2012
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Fri, 17 Aug 2012 12:35:17 -0430
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
Message-ID: <502E79CD.3070504@anew.com.ve>

Hi,

As you see in the other mails, mod_callcenter is a good option. If you 
are  an experienced programmers in Java/J2EE, you could take a look to 
mod_java (or FS javascript support) + event_socket  to build a custom 
GUI for your agents.


On 8/16/2012 7:10 AM, Nitin Tomer wrote:
>
> Hi,
>
> I am working on a Contact Center solution. It will support mail, chat 
> and call queries.
>
> The requirements are:
>
> 1. An end-customer calls, the call is handled by FreeSWITCH Auto 
> Attendant.
> 2. Customer is presented with a menu and makes selection. His call is 
> put on hold and an entry is made in my system's database for incoming 
> queries.
> 3. These queries are shown to agents handling calls.
> 4. And agent clicks on a query, he is shown an extension where call is 
> parked. He dials that and is connected to the customer.
>
> 5. He talks to the customer and resolve his queries.
>
> Please guide me on how can do it. The application will be written in Java.
>
> I am an experienced programmers in Java/J2EE; but doesn't have much 
> knowledge of VoIP/FreeSWITCH. I've configured FreeSWITCH on Ubuntu 
> 12.04; and have installed X-Lite Softphones on two Windows machines. I 
> configured these phones to work with FreeSWITCH and they are working fine.
>
> I think that I'd have to use Valet Parking to park customer's call on 
> an extension, then pass that extension to agent who will dial it and 
> will be connected to the customer.
>
> Please tell me whether this approach will work? And how should I go 
> about it.
>
> All help would be much appreciated.
>
> Regards
>
> Nitin
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2180 / Virus Database: 2437/5203 - Release Date: 08/15/12
>


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

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Tel?fono: +58 212 9081842
Cel: +58 412 2221515
dgarcia at anew.com.ve
http://www.anew.com.ve
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From brett at launch3.net  Fri Aug 17 21:16:49 2012
From: brett at launch3.net (Brett Wilson)
Date: Fri, 17 Aug 2012 13:16:49 -0400
Subject: [Freeswitch-users] mod_portaudio cannot find input or
	output	device
In-Reply-To: <CALn233qPLVkk3efB+VPeWZgw_0ziqOWTOo137oTidnPFJg1UsQ@mail.gmail.com>
References: <021301cd7c72$6adff2f0$409fd8d0$@launch3.net>	<502E3CE7.4070902@puzzled.xs4all.nl>
	<CALn233qPLVkk3efB+VPeWZgw_0ziqOWTOo137oTidnPFJg1UsQ@mail.gmail.com>
Message-ID: <01de01cd7c9c$169972f0$43cc58d0$@launch3.net>

The device(s) list fine with aplay -l and also as I mentioned sound files do
play out the speaker with aplay. When I built portaudio standalone from
portaudio's website, pa_devs also lists the sound devices fine. I tried
running FS as root and still same issue.

*******************************************?
Brett Wilson
IT Department
Launch 3 Ventures, LLC?
134 Myer Street?
Hackensack, NJ 07601?
Phone:?877.878.9134
Fax:?646.536.3866?
Email:?Brett.Wilson at launch3.net
AOL IM:?Brett.Wilson at launch3.net
www.Launch3.net?
www.Launch3telecom.com?
*******************************************?


-----Original Message-----
From: Mitch Capper [mailto:mitch.capper at gmail.com] 
Sent: Friday, August 17, 2012 10:48 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_portaudio cannot find input or output
device

The first problem may be the default config you are using is going to cause
the console to load with a lot of errors that may just add confusion.
Portaudio streams are enabled but not documented by default and are throwing
atleast some of those errors (and for most
are not needed).   Try a much simple PA config like:
                <configuration name="portaudio.conf"
description="Soundcard Endpoint">
                  <settings>
                    <param name="ring-file"
value="tone_stream://%(2000,4000,440.0,480.0);loops=20"/>
                    <param name="ring-interval" value="5"/>
                    <param name="hold-file" value="$${hold_music}"/>
                    <param name="dialplan" value="XML"/>
                    <param name="sample-rate" value="24000"/>
                    <param name="codec-ms" value="20"/>
                  </settings>
                </configuration>

Then see if you can get portaudio loading, however I believe the inability
to find any audio devices is kicking the error out.  You would want to try
running aplay -l from a command line to make sure some audio devices are
listed.  In addition try running freeswitch as root without the -u flag and
ensure that its not a permissions problem.

~Mitch

On Fri, Aug 17, 2012 at 5:45 AM, Patrick Lists
<freeswitch-list at puzzled.xs4all.nl> wrote:
> On 17-08-12 14:18, Brett Wilson wrote:
> [snip]
>> *freeswitch at internal> load mod_portaudio*
>>
>> +OK Reloading XML
>>
>> -ERR [module load file routine returned an error]
>
> That ERR does not look good. Iirc I have seen that with mod_java when 
> starting FreeSWITCH. Subsequent manual loading of the mod_java module 
> went fine. But in your case you are already manually loading the module.
>
>> *2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1767 Cannot find an 
>> input device*
>>
>> *2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1777 Cannot find an 
>> output device*
>
> Did you set the input/output devices? Lot's of info on the wiki:
>
> http://wiki.freeswitch.org/wiki/FreeSwitch_Enpoint_Portaudio#List_the_
> available_devices
>
> Regards,
> Patrick
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-use
> rs
> http://www.freeswitch.org





From brett at launch3.net  Fri Aug 17 21:17:52 2012
From: brett at launch3.net (Brett Wilson)
Date: Fri, 17 Aug 2012 13:17:52 -0400
Subject: [Freeswitch-users] mod_portaudio cannot find input or
	output	device
In-Reply-To: <502E3CE7.4070902@puzzled.xs4all.nl>
References: <021301cd7c72$6adff2f0$409fd8d0$@launch3.net>
	<502E3CE7.4070902@puzzled.xs4all.nl>
Message-ID: <01e001cd7c9c$3c454f10$b4cfed30$@launch3.net>

I disabled loading of any modules besides the bare miniumum, still no dice.
I set the in/out device as #0 as shown with aplay -l. Still no good.

*******************************************?
Brett Wilson
IT Department
Launch 3 Ventures, LLC?
134 Myer Street?
Hackensack, NJ 07601?
Phone:?877.878.9134
Fax:?646.536.3866?
Email:?Brett.Wilson at launch3.net
AOL IM:?Brett.Wilson at launch3.net
www.Launch3.net?
www.Launch3telecom.com?
*******************************************?



-----Original Message-----
From: Patrick Lists [mailto:freeswitch-list at puzzled.xs4all.nl] 
Sent: Friday, August 17, 2012 8:45 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_portaudio cannot find input or output
device

On 17-08-12 14:18, Brett Wilson wrote:
[snip]
> *freeswitch at internal> load mod_portaudio*
>
> +OK Reloading XML
>
> -ERR [module load file routine returned an error]

That ERR does not look good. Iirc I have seen that with mod_java when
starting FreeSWITCH. Subsequent manual loading of the mod_java module went
fine. But in your case you are already manually loading the module.

> *2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1767 Cannot find an 
> input device*
>
> *2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1777 Cannot find an 
> output device*

Did you set the input/output devices? Lot's of info on the wiki:

http://wiki.freeswitch.org/wiki/FreeSwitch_Enpoint_Portaudio#List_the_availa
ble_devices

Regards,
Patrick






From asaad2 at gmail.com  Fri Aug 17 21:21:34 2012
From: asaad2 at gmail.com (BookBag)
Date: Fri, 17 Aug 2012 12:21:34 -0500
Subject: [Freeswitch-users] Recording problem
In-Reply-To: <502E748C.7000900@gmail.com>
References: <502E7201.5050909@gmail.com>
	<502E748C.7000900@gmail.com>
Message-ID: <CAAdTMa_n-rQay6e+OUrB1w9qYS3s017Nj1SJTBYOCB4xo9wiBg@mail.gmail.com>

So its always the same user who has good audio or does it matter who gets
the call first?
On Aug 17, 2012 12:46 PM, "Vbvbrj" <vbvbrj at gmail.com> wrote:

> Forgot to mention. The user 743 has the good record audio file, the
> other two are bad recorded audio files.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 vbvbrj at gmail.com  Fri Aug 17 21:27:34 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Fri, 17 Aug 2012 20:27:34 +0300
Subject: [Freeswitch-users] Recording problem
In-Reply-To: <CAAdTMa_n-rQay6e+OUrB1w9qYS3s017Nj1SJTBYOCB4xo9wiBg@mail.gmail.com>
References: <502E7201.5050909@gmail.com> <502E748C.7000900@gmail.com>
	<CAAdTMa_n-rQay6e+OUrB1w9qYS3s017Nj1SJTBYOCB4xo9wiBg@mail.gmail.com>
Message-ID: <502E7F06.8090307@gmail.com>

On 17.08.2012 20:21, BookBag wrote:
> So its always the same user who has good audio or does it matter who
> gets the call first?

The good record have the user 743 always, the other two always have bad 
sound record.



From mitch.capper at gmail.com  Fri Aug 17 21:39:27 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Fri, 17 Aug 2012 10:39:27 -0700
Subject: [Freeswitch-users] mod_portaudio cannot find input or output
	device
In-Reply-To: <01e001cd7c9c$3c454f10$b4cfed30$@launch3.net>
References: <021301cd7c72$6adff2f0$409fd8d0$@launch3.net>
	<502E3CE7.4070902@puzzled.xs4all.nl>
	<01e001cd7c9c$3c454f10$b4cfed30$@launch3.net>
Message-ID: <CALn233r1Y0QfHzkT7fU_N8neSztcMLhx38UR2ZnjNy_Xydiejg@mail.gmail.com>

in/outdev dont matter as portaudio is saying it cant find any device.
Paste the results of:
ps auxf | grep freeswitch

back.

~Mitch

On Fri, Aug 17, 2012 at 10:17 AM, Brett Wilson <brett at launch3.net> wrote:
> I disabled loading of any modules besides the bare miniumum, still no dice.
> I set the in/out device as #0 as shown with aplay -l. Still no good.
>
> *******************************************
> Brett Wilson
> IT Department
> Launch 3 Ventures, LLC
> 134 Myer Street
> Hackensack, NJ 07601
> Phone: 877.878.9134
> Fax: 646.536.3866
> Email: Brett.Wilson at launch3.net
> AOL IM: Brett.Wilson at launch3.net
> www.Launch3.net
> www.Launch3telecom.com
> *******************************************
>
>
>
> -----Original Message-----
> From: Patrick Lists [mailto:freeswitch-list at puzzled.xs4all.nl]
> Sent: Friday, August 17, 2012 8:45 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_portaudio cannot find input or output
> device
>
> On 17-08-12 14:18, Brett Wilson wrote:
> [snip]
>> *freeswitch at internal> load mod_portaudio*
>>
>> +OK Reloading XML
>>
>> -ERR [module load file routine returned an error]
>
> That ERR does not look good. Iirc I have seen that with mod_java when
> starting FreeSWITCH. Subsequent manual loading of the mod_java module went
> fine. But in your case you are already manually loading the module.
>
>> *2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1767 Cannot find an
>> input device*
>>
>> *2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1777 Cannot find an
>> output device*
>
> Did you set the input/output devices? Lot's of info on the wiki:
>
> http://wiki.freeswitch.org/wiki/FreeSwitch_Enpoint_Portaudio#List_the_availa
> ble_devices
>
> Regards,
> Patrick
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 info at pripojtese.net  Fri Aug 17 20:17:39 2012
From: info at pripojtese.net (Jakub Tencl)
Date: Fri, 17 Aug 2012 17:17:39 +0100
Subject: [Freeswitch-users] IP trunk without registration to the carrier
In-Reply-To: <CAKzWOxXhnxSjPphJ9Zm-KeJFFWBEtt6QK2wkix0xRTLbB9OwVA@mail.gmail.com>
References: <001401cd7c62$3c1f35d0$b45da170$@com>
	<CAKzWOxXhnxSjPphJ9Zm-KeJFFWBEtt6QK2wkix0xRTLbB9OwVA@mail.gmail.com>
Message-ID: <502E6EA3.4090104@pripojtese.net>

Hi,

i am looking for posibility how to assign the carrier without 
authentification not user i mean between freeswitch and my provider, can 
you help me please?

Thanks





From anton.jugatsu at gmail.com  Fri Aug 17 21:58:05 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Fri, 17 Aug 2012 21:58:05 +0400
Subject: [Freeswitch-users] IP trunk without registration to the carrier
In-Reply-To: <502E6EA3.4090104@pripojtese.net>
References: <001401cd7c62$3c1f35d0$b45da170$@com>
	<CAKzWOxXhnxSjPphJ9Zm-KeJFFWBEtt6QK2wkix0xRTLbB9OwVA@mail.gmail.com>
	<502E6EA3.4090104@pripojtese.net>
Message-ID: <CANBhyPDMVoq1b2NCqf=ETi9AXQzapdr2uow8CSn8B-jqDGmUEQ@mail.gmail.com>

You can you ACL, search wiki.

2012/8/17 Jakub Tencl <info at pripojtese.net>

> Hi,
>
> i am looking for posibility how to assign the carrier without
> authentification not user i mean between freeswitch and my provider, can
> you help me please?
>
> Thanks
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 asaad2 at gmail.com  Fri Aug 17 23:11:05 2012
From: asaad2 at gmail.com (BookBag)
Date: Fri, 17 Aug 2012 15:11:05 -0400
Subject: [Freeswitch-users] Recording problem
In-Reply-To: <502E7F06.8090307@gmail.com>
References: <502E7201.5050909@gmail.com> <502E748C.7000900@gmail.com>
	<CAAdTMa_n-rQay6e+OUrB1w9qYS3s017Nj1SJTBYOCB4xo9wiBg@mail.gmail.com>
	<502E7F06.8090307@gmail.com>
Message-ID: <CAAdTMa8DdeRCdYGRKeD2qJLAZyfLr9Sv08-R=_YMNSaSH_svVQ@mail.gmail.com>

Sounds like network connection. Can you try switching stations between a
bad phone and a good phone. Maybe your network has bad ports or cables
somewhere. I know for all cisco voip they recommend cat 6 cables.
On Aug 17, 2012 1:32 PM, "Vbvbrj" <vbvbrj at gmail.com> wrote:

> On 17.08.2012 20:21, BookBag wrote:
> > So its always the same user who has good audio or does it matter who
> > gets the call first?
>
> The good record have the user 743 always, the other two always have bad
> sound record.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 brett at launch3.net  Sat Aug 18 00:42:35 2012
From: brett at launch3.net (Brett Wilson)
Date: Fri, 17 Aug 2012 16:42:35 -0400
Subject: [Freeswitch-users] mod_portaudio cannot find input or
	output	device
In-Reply-To: <CALn233r1Y0QfHzkT7fU_N8neSztcMLhx38UR2ZnjNy_Xydiejg@mail.gmail.com>
References: <021301cd7c72$6adff2f0$409fd8d0$@launch3.net>	<502E3CE7.4070902@puzzled.xs4all.nl>	<01e001cd7c9c$3c454f10$b4cfed30$@launch3.net>
	<CALn233r1Y0QfHzkT7fU_N8neSztcMLhx38UR2ZnjNy_Xydiejg@mail.gmail.com>
Message-ID: <020301cd7cb8$d5985790$80c906b0$@launch3.net>

root     16791  1.6  0.9  37076 14456 ?        S<l  16:33   0:08
/usr/local/freeswitch/bin/freeswitch -rp -nc

*******************************************?
Brett Wilson
IT Department
Launch 3 Ventures, LLC?
134 Myer Street?
Hackensack, NJ 07601?
Phone:?877.878.9134
Fax:?646.536.3866?
Email:?Brett.Wilson at launch3.net
AOL IM:?Brett.Wilson at launch3.net
www.Launch3.net?
www.Launch3telecom.com?
*******************************************?



-----Original Message-----
From: Mitch Capper [mailto:mitch.capper at gmail.com] 
Sent: Friday, August 17, 2012 1:39 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_portaudio cannot find input or output
device

in/outdev dont matter as portaudio is saying it cant find any device.
Paste the results of:
ps auxf | grep freeswitch

back.

~Mitch

On Fri, Aug 17, 2012 at 10:17 AM, Brett Wilson <brett at launch3.net> wrote:
> I disabled loading of any modules besides the bare miniumum, still no
dice.
> I set the in/out device as #0 as shown with aplay -l. Still no good.
>
> *******************************************
> Brett Wilson
> IT Department
> Launch 3 Ventures, LLC
> 134 Myer Street
> Hackensack, NJ 07601
> Phone: 877.878.9134
> Fax: 646.536.3866
> Email: Brett.Wilson at launch3.net
> AOL IM: Brett.Wilson at launch3.net
> www.Launch3.net
> www.Launch3telecom.com
> *******************************************
>
>
>
> -----Original Message-----
> From: Patrick Lists [mailto:freeswitch-list at puzzled.xs4all.nl]
> Sent: Friday, August 17, 2012 8:45 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_portaudio cannot find input or 
> output device
>
> On 17-08-12 14:18, Brett Wilson wrote:
> [snip]
>> *freeswitch at internal> load mod_portaudio*
>>
>> +OK Reloading XML
>>
>> -ERR [module load file routine returned an error]
>
> That ERR does not look good. Iirc I have seen that with mod_java when 
> starting FreeSWITCH. Subsequent manual loading of the mod_java module 
> went fine. But in your case you are already manually loading the module.
>
>> *2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1767 Cannot find an 
>> input device*
>>
>> *2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1777 Cannot find an 
>> output device*
>
> Did you set the input/output devices? Lot's of info on the wiki:
>
> http://wiki.freeswitch.org/wiki/FreeSwitch_Enpoint_Portaudio#List_the_
> availa
> ble_devices
>
> Regards,
> Patrick
>
>
>
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-use
> rs
> http://www.freeswitch.org





From mailing-lists at phoenixinternet.net  Sat Aug 18 02:58:19 2012
From: mailing-lists at phoenixinternet.net (Gilbert T. Gutierrez, Jr.)
Date: Fri, 17 Aug 2012 15:58:19 -0700
Subject: [Freeswitch-users] mod_python
Message-ID: <502ECC8B.204@phoenixinternet.net>

Hello List!
You will have to excuse me if this has been answered before, but I am 
having an issue compiling mod_python. At the end of my message is my 
output. I am running Centos 6 x64 and following the instructions from 
FreePyBX (I want to see and test the interface). I have successfully 
compiled FreeSwitch on Centos6 before but in this instance, FreePyBX 
requests that I use Python 2.7.3 in their instructions. I have even 
tried including '--enable-shared' in my Python compilation with the same 
results when I compile FreeSwitch. Can someone help me on getting past 
this error?

Thank you,
Gilbert T. Gutierrez, Jr.

making all mod_python
Compiling freeswitch_python.cpp...
cc1plus: warning: command line option "-Wstrict-prototypes" is valid for 
Ada/C/ObjC but not for C++
In file included from /usr/include/python2.7/Python.h:8,
                  from freeswitch_python.h:5,
                  from freeswitch_python.cpp:3:
/usr/include/python2.7/pyconfig.h:1161:1: warning: "_POSIX_C_SOURCE" 
redefined
In file included from /usr/include/stdlib.h:25,
                  from /usr/local/src/freeswitch/src/include/switch.h:76,
                  from freeswitch_python.cpp:2:
/usr/include/features.h:162:1: warning: this is the location of the 
previous definition
In file included from /usr/include/python2.7/Python.h:8,
                  from freeswitch_python.h:5,
                  from freeswitch_python.cpp:3:
/usr/include/python2.7/pyconfig.h:1183:1: warning: "_XOPEN_SOURCE" redefined
In file included from /usr/include/stdlib.h:25,
                  from /usr/local/src/freeswitch/src/include/switch.h:76,
                  from freeswitch_python.cpp:2:
/usr/include/features.h:164:1: warning: this is the location of the 
previous definition
freeswitch_python.cpp: In member function ?virtual switch_status_t 
PYTHON::Session::run_dtmf_callback(void*, switch_input_type_t)?:
freeswitch_python.cpp:287: warning: deprecated conversion from string 
constant to ?char*?
freeswitch_python.cpp:302: warning: deprecated conversion from string 
constant to ?char*?
freeswitch_python.cpp:304: warning: deprecated conversion from string 
constant to ?char*?
Compiling mod_python_wrap.cpp...
cc1plus: warning: command line option "-Wstrict-prototypes" is valid for 
Ada/C/ObjC but not for C++
mod_python_wrap.cpp: In function ?PyObject* _wrap_new_Stream(PyObject*, 
PyObject*)?:
mod_python_wrap.cpp:4178: warning: ?argv[0]? may be used uninitialized 
in this function
Compiling 
/usr/local/src/freeswitch/src/mod/languages/mod_python/mod_python.c...
quiet_libtool: compile:  gcc -I/usr/include/python2.7 
-I/usr/include/python2.7 -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv 
-O3 -Wall -Wstrict-prototypes 
-I/usr/local/src/freeswitch/libs/curl/include 
-I/usr/local/src/freeswitch/src/include 
-I/usr/local/src/freeswitch/src/include 
-I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror 
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g 
-ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic 
-Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c 
/usr/local/src/freeswitch/src/mod/languages/mod_python/mod_python.c 
-fPIC -DPIC -o .libs/mod_python.o
quiet_libtool: compile:  gcc -I/usr/include/python2.7 
-I/usr/include/python2.7 -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv 
-O3 -Wall -Wstrict-prototypes 
-I/usr/local/src/freeswitch/libs/curl/include 
-I/usr/local/src/freeswitch/src/include 
-I/usr/local/src/freeswitch/src/include 
-I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror 
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g 
-ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic 
-Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c 
/usr/local/src/freeswitch/src/mod/languages/mod_python/mod_python.c -o 
mod_python.o >/dev/null 2>&1
Creating mod_python.la...
/usr/bin/ld: 
/usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../../libpython2.7.a(abstract.o): 
relocation R_X86_64_32 against `.rodata.str1.8' can not be used when 
making a shared object; recompile with -fPIC
/usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../../libpython2.7.a: could 
not read symbols: Bad value
collect2: ld returned 1 exit status
cat: .libs/mod_python.log: No such file or directory
make[5]: *** [mod_python.la] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_python-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2





From mailing-lists at phoenixinternet.net  Sat Aug 18 03:52:55 2012
From: mailing-lists at phoenixinternet.net (Gilbert T. Gutierrez, Jr.)
Date: Fri, 17 Aug 2012 16:52:55 -0700
Subject: [Freeswitch-users] mod_python
In-Reply-To: <502ECC8B.204@phoenixinternet.net>
References: <502ECC8B.204@phoenixinternet.net>
Message-ID: <502ED957.5010508@phoenixinternet.net>

Actually I got it to work... I had to recompile Python for the 100th 
time and add '-fPIC' to the "CC=" line (the line in my Python make file 
reads "CC= gcc -pthread -fPIC"). Evidently the "--enable-shared" does 
not always work (I found someone else with Ubuntu 64 bit having a 
similar issue with Python).

Gilbert

On 8/17/2012 3:58 PM, Gilbert T. Gutierrez, Jr. wrote:
> Hello List!
> You will have to excuse me if this has been answered before, but I am
> having an issue compiling mod_python. At the end of my message is my
> output. I am running Centos 6 x64 and following the instructions from
> FreePyBX (I want to see and test the interface). I have successfully
> compiled FreeSwitch on Centos6 before but in this instance, FreePyBX
> requests that I use Python 2.7.3 in their instructions. I have even
> tried including '--enable-shared' in my Python compilation with the same
> results when I compile FreeSwitch. Can someone help me on getting past
> this error?
>
> Thank you,
> Gilbert T. Gutierrez, Jr.
>
> making all mod_python
> Compiling freeswitch_python.cpp...
> cc1plus: warning: command line option "-Wstrict-prototypes" is valid for
> Ada/C/ObjC but not for C++
> In file included from /usr/include/python2.7/Python.h:8,
>                    from freeswitch_python.h:5,
>                    from freeswitch_python.cpp:3:
> /usr/include/python2.7/pyconfig.h:1161:1: warning: "_POSIX_C_SOURCE"
> redefined
> In file included from /usr/include/stdlib.h:25,
>                    from /usr/local/src/freeswitch/src/include/switch.h:76,
>                    from freeswitch_python.cpp:2:
> /usr/include/features.h:162:1: warning: this is the location of the
> previous definition
> In file included from /usr/include/python2.7/Python.h:8,
>                    from freeswitch_python.h:5,
>                    from freeswitch_python.cpp:3:
> /usr/include/python2.7/pyconfig.h:1183:1: warning: "_XOPEN_SOURCE" redefined
> In file included from /usr/include/stdlib.h:25,
>                    from /usr/local/src/freeswitch/src/include/switch.h:76,
>                    from freeswitch_python.cpp:2:
> /usr/include/features.h:164:1: warning: this is the location of the
> previous definition
> freeswitch_python.cpp: In member function ?virtual switch_status_t
> PYTHON::Session::run_dtmf_callback(void*, switch_input_type_t)?:
> freeswitch_python.cpp:287: warning: deprecated conversion from string
> constant to ?char*?
> freeswitch_python.cpp:302: warning: deprecated conversion from string
> constant to ?char*?
> freeswitch_python.cpp:304: warning: deprecated conversion from string
> constant to ?char*?
> Compiling mod_python_wrap.cpp...
> cc1plus: warning: command line option "-Wstrict-prototypes" is valid for
> Ada/C/ObjC but not for C++
> mod_python_wrap.cpp: In function ?PyObject* _wrap_new_Stream(PyObject*,
> PyObject*)?:
> mod_python_wrap.cpp:4178: warning: ?argv[0]? may be used uninitialized
> in this function
> Compiling
> /usr/local/src/freeswitch/src/mod/languages/mod_python/mod_python.c...
> quiet_libtool: compile:  gcc -I/usr/include/python2.7
> -I/usr/include/python2.7 -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv
> -O3 -Wall -Wstrict-prototypes
> -I/usr/local/src/freeswitch/libs/curl/include
> -I/usr/local/src/freeswitch/src/include
> -I/usr/local/src/freeswitch/src/include
> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g
> -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic
> -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c
> /usr/local/src/freeswitch/src/mod/languages/mod_python/mod_python.c
> -fPIC -DPIC -o .libs/mod_python.o
> quiet_libtool: compile:  gcc -I/usr/include/python2.7
> -I/usr/include/python2.7 -fno-strict-aliasing -g -O2 -DNDEBUG -g -fwrapv
> -O3 -Wall -Wstrict-prototypes
> -I/usr/local/src/freeswitch/libs/curl/include
> -I/usr/local/src/freeswitch/src/include
> -I/usr/local/src/freeswitch/src/include
> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g
> -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic
> -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c
> /usr/local/src/freeswitch/src/mod/languages/mod_python/mod_python.c -o
> mod_python.o >/dev/null 2>&1
> Creating mod_python.la...
> /usr/bin/ld:
> /usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../../libpython2.7.a(abstract.o):
> relocation R_X86_64_32 against `.rodata.str1.8' can not be used when
> making a shared object; recompile with -fPIC
> /usr/lib/gcc/x86_64-redhat-linux/4.4.6/../../../libpython2.7.a: could
> not read symbols: Bad value
> collect2: ld returned 1 exit status
> cat: .libs/mod_python.log: No such file or directory
> make[5]: *** [mod_python.la] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_python-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 obiebrown at gmail.com  Sat Aug 18 04:44:41 2012
From: obiebrown at gmail.com (Obie Brown)
Date: Sat, 18 Aug 2012 10:14:41 +0930
Subject: [Freeswitch-users] Recording problem
In-Reply-To: <CAAdTMa8DdeRCdYGRKeD2qJLAZyfLr9Sv08-R=_YMNSaSH_svVQ@mail.gmail.com>
References: <502E7201.5050909@gmail.com> <502E748C.7000900@gmail.com>
	<CAAdTMa_n-rQay6e+OUrB1w9qYS3s017Nj1SJTBYOCB4xo9wiBg@mail.gmail.com>
	<502E7F06.8090307@gmail.com>
	<CAAdTMa8DdeRCdYGRKeD2qJLAZyfLr9Sv08-R=_YMNSaSH_svVQ@mail.gmail.com>
Message-ID: <CA+aeUL8j2hwmQ+wyf8YZj3d-60A5TK4BUjvkYi58KJ=1+OWiYw@mail.gmail.com>

I am very new to FreeSWITCH but I have been using it. Also I used Newfies
and found the same problem.

I just re sampled all my recordings to WAV in 8000Hz format and it fixed
the problem. I do not know much about things but could this maybe be a
codec problem in that your agents recordings that "work" are connecting to
system with XXX codec yet your other agent that is "not working" are
connecting with a different codec?

Thanks,
Obie Brown
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From cmason at frontiernetworks.ca  Sat Aug 18 05:22:44 2012
From: cmason at frontiernetworks.ca (Colin Mason)
Date: Fri, 17 Aug 2012 21:22:44 -0400
Subject: [Freeswitch-users] sofia_contact changes in new git
Message-ID: <0D1C698866F66045A6201FD0F59CAC9001465874EC@EX.frontier.local>

Hello,

I'm using a single domain freeswitch box with 2 sip_profiles used to register endpoints.

I've noticed a change in how "user/" and "sofia_contact" work in the latest gits that have been commited. I used to be able to go into the freeswitch console and type:


-          sofia_contact 1001
reply:    sofia/mpls/sip:1001 at 10.253.200.111:5060


If I use the above command in the latest gits I get: "error/user_not_registered"
So instead I have to type:


-          sofia_contact mpls/1001
reply:    sofia/mpls/sip:1001 at 10.253.200.111:5060


-          sofia_contact */1001
reply:                sofia/mpls/sip:1001 at 10.253.200.111:5060,sofia/mpls/sip:1001 at 10.253.200.111:5060,sofia/mpls/sip:1001 at 10.253.200.111:5060


The first command works on:

FreeSWITCH Version 1.2.0-rc2+git~20120723T163619Z~524468be7b (1.2.0-rc2; git at commit 524468be7b on Mon, 23 Jul 2012 16:36:19 Z)

But is broken on:

FreeSWITCH Version 1.2.1+git~20120814T055744Z~f2f4f4acff (1.2.1; git at commit f2f4f4acff on Tue, 14 Aug 2012 05:57:44 Z)
FreeSWITCH Version 1.2.0-rc2+git~20120817T190720Z~8c6b8edfea (1.2.0-rc2; git at commit 8c6b8edfea on Fri, 17 Aug 2012 19:07:20 Z)

I am using identical configurations. I was wondering if this was intended behavior and if there is some sort of workaround to get freeswitch/sofia_contact to check which profile an extension is registered on in these recent git commits.  Currently if I try to bridge to "user/1001" it fails and group_call is also failing because of this change whereas they work perfectly with the old git. Thanks in advance.

Colin


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From stephen at picardogroup.com  Sat Aug 18 06:08:54 2012
From: stephen at picardogroup.com (stephen at picardogroup.com)
Date: Fri, 17 Aug 2012 19:08:54 -0700
Subject: [Freeswitch-users] Audio Delay
Message-ID: <20120817190854.0e1bd4d5c5064b420440751b21b10e46.b4df44975c.wbe@email13.secureserver.net>

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From vbvbrj at gmail.com  Sat Aug 18 10:19:12 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Sat, 18 Aug 2012 09:19:12 +0300
Subject: [Freeswitch-users] Recording problem
In-Reply-To: <CAAdTMa8DdeRCdYGRKeD2qJLAZyfLr9Sv08-R=_YMNSaSH_svVQ@mail.gmail.com>
References: <502E7201.5050909@gmail.com> <502E748C.7000900@gmail.com>
	<CAAdTMa_n-rQay6e+OUrB1w9qYS3s017Nj1SJTBYOCB4xo9wiBg@mail.gmail.com>
	<502E7F06.8090307@gmail.com>
	<CAAdTMa8DdeRCdYGRKeD2qJLAZyfLr9Sv08-R=_YMNSaSH_svVQ@mail.gmail.com>
Message-ID: <502F33E0.5020800@gmail.com>

On 17.08.2012 22:11, BookBag wrote:
> Sounds like network connection. Can you try switching stations between a
> bad phone and a good phone. Maybe your network has bad ports or cables
> somewhere. I know for all cisco voip they recommend cat 6 cables.

It is definatelly not a network problem. The clients are connected to 
the same switch. I posted the log to see the difference. Observe "codec 
substitution" lines for bad clients. I don't know what this means to the 
system as why a substitution is done, but are not tried other codecs 
from client to compare.


From boris at tagnet.ru  Sat Aug 18 11:35:39 2012
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 18 Aug 2012 13:35:39 +0600
Subject: [Freeswitch-users] make current and 1.2
Message-ID: <502F45CB.1050204@tagnet.ru>

Hello!

     1.2 released. But after "make current" I get:
freeswitch at default> version
FreeSWITCH Version 1.2.0-rc2+git~20120818T004654Z~bdb73beb8e (1.2.0-rc2; 
git at commit bdb73beb8e on Sat, 18 Aug 2012 00:46:54 Z)

What am I doing wrong?

-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005
   http://www.tagnet.ru



From peter.olsson at visionutveckling.se  Sat Aug 18 11:47:45 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 18 Aug 2012 07:47:45 +0000
Subject: [Freeswitch-users] make current and 1.2
In-Reply-To: <502F45CB.1050204@tagnet.ru>
References: <502F45CB.1050204@tagnet.ru>
Message-ID: <06068E38-4022-4551-8EEB-39AE6427BF61@visionutveckling.se>

The master branch is still set to this version. If you want to stick to the 1.2 stable version, change branch to 1.2.stable (I think that's the correct name).

/Peter

18 aug 2012 kl. 09:44 skrev "Boris Kovalenko" <boris at tagnet.ru>:

> Hello!
> 
>     1.2 released. But after "make current" I get:
> freeswitch at default> version
> FreeSWITCH Version 1.2.0-rc2+git~20120818T004654Z~bdb73beb8e (1.2.0-rc2; 
> git at commit bdb73beb8e on Sat, 18 Aug 2012 00:46:54 Z)
> 
> What am I doing wrong?
> 
> -- 
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435) 230001
>   ???? +7 (3435) 230005
>   http://www.tagnet.ru
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> !DSPAM:502f443b32761991087014!
> 


From krice at freeswitch.org  Sat Aug 18 19:16:46 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 18 Aug 2012 10:16:46 -0500
Subject: [Freeswitch-users] make current and 1.2
In-Reply-To: <06068E38-4022-4551-8EEB-39AE6427BF61@visionutveckling.se>
Message-ID: <CC551C0E.3C31B%krice@freeswitch.org>

The stable tracking branch is v1.2.stable

We should probably wiki up instructions for make current on that


On 8/18/12 2:47 AM, "Peter Olsson" <peter.olsson at visionutveckling.se> wrote:

> The master branch is still set to this version. If you want to stick to the
> 1.2 stable version, change branch to 1.2.stable (I think that's the correct
> name).

/Peter

18 aug 2012 kl. 09:44 skrev "Boris Kovalenko"
> <boris at tagnet.ru>:

> Hello!
> 
>     1.2 released. But after "make current" I
> get:
> freeswitch at default> version
> FreeSWITCH Version
> 1.2.0-rc2+git~20120818T004654Z~bdb73beb8e (1.2.0-rc2; 
> git at commit
> bdb73beb8e on Sat, 18 Aug 2012 00:46:54 Z)
> 
> What am I doing wrong?
> 
> --
> 
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435)
> 230001
>   ???? +7 (3435) 230005
>   http://www.tagnet.ru
> 
> 
>
> _________________________________________________________________________>
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
> 
> FreeSWITCH-powered IP PBX: The CudaTel
> Communication Server
> 
> 
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
> http://wiki.freeswitch.org
>
> http://www.cluecon.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
> 
> !DSPAM:502f443b32761991087014!
>
> 

_________________________________________________________________________
Pr
> ofessional FreeSWITCH Consulting
> Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

FreeSW
> ITCH-powered IP PBX: The CudaTel Communication
> Server


Official FreeSWITCH
> Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.
> 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/opt
> ions/freeswitch-users
http://www.freeswitch.org


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From andrew at cassidywebservices.co.uk  Sun Aug 19 00:56:01 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sat, 18 Aug 2012 21:56:01 +0100
Subject: [Freeswitch-users] mobile diallers
In-Reply-To: <26f073599ffd326824ba44e1ec40fa4f@www.webmail.co.za>
References: <mailman.17045.1345189578.13443.freeswitch-users@lists.freeswitch.org>
	<26f073599ffd326824ba44e1ec40fa4f@www.webmail.co.za>
Message-ID: <CAMbW9oLUBzXJSZk0x21fCACiZe5VWrDLUE8hwg9isgvb2yuEGQ@mail.gmail.com>

I've tried acrobits (slightly), SipDroid, CSipSimple and the built-in
android client in gingerbread. They're best avoided for incoming calls if
you value your battery life, enabling registration for inbound calls can
reduce your battery life rather drastically.

On 17 August 2012 09:24, clive engelberg <clive18 at webmail.co.za> wrote:

> Hi
>
>
>
> Someone recommended http://www.acrobits.cz
>
>
>
> Worth a try.
>
>
>
> regards
>
> Clive
>
>
>
> On Fri, 17 Aug 2012 11:46:18 +0400
> freeswitch-users-request at lists.freeswitch.org wrote
>
>  IMSDroid is based on Doubango developed by Doubango Telecom, which also
> has support for iPhone, Windows and Mac.
> You should check their website at http://www.doubango.org. I think they
> may provide you commercial support for their productions as well, just talk
> to them.
> Thank you.
>
>
> On Fri, Aug 17, 2012 at 9:05 AM, vaad.fabi at gmail.com <vaad.fabi at gmail.com>wrote:
>
>>
>> IMSDROID - only Android platform, no  centralizied dev team\vendor with
>> support and SLA
>> SIPDROID - only Android platform, no  centralizied dev team\vendor with
>> support and SLA
>> Counterpath - as i know no mobile platforms
>>
>> Searching...
>>
>>
>>
>> On 08/16/2012 08:58 PM, freeswitch wrote:
>>
>>  IMSDROID will be good!
>>
>>
>>  At 2012-08-16 20:44:54,BookBag <asaad2 at gmail.com> <asaad2 at gmail.com>wrote:
>>
>> I'm surprised no one said Bria by counterpath. Which allows for xmpp and
>> presence. The softphone allows you to send sms also if you buy the full
>> version. Another one that I tested is Csipsimple, this one allows sms for
>> free. But my absolute favorite is Zoiper, unfortunately no sms capability
>> but wasnt buggy except for one bug I encountered with sending the correct
>> dtmf style. Thank god their new update took care of that. Phone now seems
>> very solid and now includes a QR reader.
>>
>> On Thu, Aug 16, 2012 at 6:16 AM, vaad.fabi at gmail.com <vaad.fabi at gmail.com
>> > wrote:
>>
>>>   Does they provide these software development, customisation and
>>> support ? i think no.  I need vendor\development team who can provide
>>> development\customisation\support.
>>> Anyway thx.
>>>
>>>
>>> On 08/16/2012 12:53 PM, Mohammad Emran wrote:
>>> > Then try nimbuzz or fring.... If it works for u.
>>> >
>>> > Sent from my iPhone
>>> >
>>> > On Aug 16, 2012, at 3:42 PM, "vaad.fabi at gmail.com"<vaad.fabi at gmail.com>
>>>  wrote:
>>> >
>>> >>   Seems to be ok for a simple dialer, but i forgot to mention - i need
>>> >> advanced options for dialers, such as: IM (xmpp\facebook\twitter
>>> >> integration), video calls, balance state, check direction rate,
>>> >> callback, sms, top up and etc. It's smth like UC application with
>>> >> calling support as one of the many options.
>>> >>
>>> >>
>>> >> On 08/16/2012 12:22 PM, Mohammad Emran wrote:
>>> >>> Try www.sipmobiledialer.com
>>> >>>
>>> >>>
>>> >>> Sent from my iPad
>>> >>>
>>> >>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com"<
>>> vaad.fabi at gmail.com>   wrote:
>>> >>>
>>> >>>>   Hi all,
>>> >>>>
>>> >>>> Please suggest mobile voip dialer vendor for
>>> >>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>>> >>>> don't suggest google plz
>>> >>>>
>>> >>>> --
>>> >>>> Best Regards,
>>> >>>> Vadim F.
>>> >>>>
>>> >>>>
>>> >>>>
>>> _________________________________________________________________________
>>> >>>> Professional FreeSWITCH Consulting Services:
>>> >>>> consulting at freeswitch.org
>>> >>>> http://www.freeswitchsolutions.com
>>> >>>>
>>> >>>> 
>>> >>>> 
>>> >>>>
>>> >>>> Official FreeSWITCH Sites
>>> >>>> http://www.freeswitch.org
>>> >>>> http://wiki.freeswitch.org
>>> >>>> http://www.cluecon.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
>>> >>>
>>> _________________________________________________________________________
>>> >>> Professional FreeSWITCH Consulting Services:
>>> >>> consulting at freeswitch.org
>>> >>> http://www.freeswitchsolutions.com
>>> >>>
>>> >>> 
>>> >>> 
>>> >>>
>>> >>> Official FreeSWITCH Sites
>>> >>> http://www.freeswitch.org
>>> >>> http://wiki.freeswitch.org
>>> &g t;>> http://www.cluecon.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
>>> >>
>>> >> --
>>> >> Best Regards,
>>> >> Vadim F.
>>> >>
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> co nsulting at freeswitch.org <consulting at freeswitch.org>
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> 
>>> >> 
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://wiki.freeswitch.org
>>> >> http://www.cluecon.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://lists.freeswitch.org/mailman/optio%20ns/freeswitch-users>
>>> >> http://www.freeswitch.org
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.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
>>>
>>>
>>> --
>>> Best Regards,
>>> Vadim F.
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>> < br /> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>>
>>  _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com     Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com  FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.free
>>  switch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org
>>
>>
>>
>> --  Best Regards, Vadim F.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSU BSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> ------------------------------
> South Africa premier free email service - webmail.co.za<http://www.webmail.co.za/>
> <http://b.wm.co.za/click.pwm?cid=20050843&loc=N-MT&seq=502dffd1>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From shaheryarkh at googlemail.com  Sun Aug 19 01:04:53 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Sat, 18 Aug 2012 23:04:53 +0200
Subject: [Freeswitch-users] mobile diallers
In-Reply-To: <CAMbW9oLUBzXJSZk0x21fCACiZe5VWrDLUE8hwg9isgvb2yuEGQ@mail.gmail.com>
References: <mailman.17045.1345189578.13443.freeswitch-users@lists.freeswitch.org>
	<26f073599ffd326824ba44e1ec40fa4f@www.webmail.co.za>
	<CAMbW9oLUBzXJSZk0x21fCACiZe5VWrDLUE8hwg9isgvb2yuEGQ@mail.gmail.com>
Message-ID: <CAFZQphz6VSFv0EGDGchJ0AYZ+=nRh3=disz-u1WjDDsnVf4CbA@mail.gmail.com>

Yup, that's correct and the solution on Android platform is GCM (Google
Cloud Messaging) and on iPhone APNS (Apple Push Notification Service). Such
that whenever there is an incoming call for the device, the server sends a
push notification to device to intimate an incoming call, the device
suppose to active SIP client, which immediately gets SIP Registered and as
soon as server see client has come online it sends call to it. Then as soon
as call finishes SIP client de-register itself from SIP server and exits.
This considerably saves battery and data overheads.

Thank you.


On Sat, Aug 18, 2012 at 10:56 PM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> I've tried acrobits (slightly), SipDroid, CSipSimple and the built-in
> android client in gingerbread. They're best avoided for incoming calls if
> you value your battery life, enabling registration for inbound calls can
> reduce your battery life rather drastically.
>
>
> On 17 August 2012 09:24, clive engelberg <clive18 at webmail.co.za> wrote:
>
>> Hi
>>
>>
>>
>> Someone recommended http://www.acrobits.cz
>>
>>
>>
>> Worth a try.
>>
>>
>>
>> regards
>>
>> Clive
>>
>>
>>
>> On Fri, 17 Aug 2012 11:46:18 +0400
>> freeswitch-users-request at lists.freeswitch.org wrote
>>
>>  IMSDroid is based on Doubango developed by Doubango Telecom, which also
>> has support for iPhone, Windows and Mac.
>> You should check their website at http://www.doubango.org. I think they
>> may provide you commercial support for their productions as well, just talk
>> to them.
>> Thank you.
>>
>>
>> On Fri, Aug 17, 2012 at 9:05 AM, vaad.fabi at gmail.com <vaad.fabi at gmail.com
>> > wrote:
>>
>>>
>>> IMSDROID - only Android platform, no  centralizied dev team\vendor with
>>> support and SLA
>>> SIPDROID - only Android platform, no  centralizied dev team\vendor with
>>> support and SLA
>>> Counterpath - as i know no mobile platforms
>>>
>>> Searching...
>>>
>>>
>>>
>>> On 08/16/2012 08:58 PM, freeswitch wrote:
>>>
>>>  IMSDROID will be good!
>>>
>>>
>>>  At 2012-08-16 20:44:54,BookBag <asaad2 at gmail.com> <asaad2 at gmail.com>wrote:
>>>
>>> I'm surprised no one said Bria by counterpath. Which allows for xmpp and
>>> presence. The softphone allows you to send sms also if you buy the full
>>> version. Another one that I tested is Csipsimple, this one allows sms for
>>> free. But my absolute favorite is Zoiper, unfortunately no sms capability
>>> but wasnt buggy except for one bug I encountered with sending the correct
>>> dtmf style. Thank god their new update took care of that. Phone now seems
>>> very solid and now includes a QR reader.
>>>
>>> On Thu, Aug 16, 2012 at 6:16 AM, vaad.fabi at gmail.com <
>>> vaad.fabi at gmail.com> wrote:
>>>
>>>>   Does they provide these software development, customisation and
>>>> support ? i think no.  I need vendor\development team who can provide
>>>> development\customisation\support.
>>>> Anyway thx.
>>>>
>>>>
>>>> On 08/16/2012 12:53 PM, Mohammad Emran wrote:
>>>> > Then try nimbuzz or fring.... If it works for u.
>>>> >
>>>> > Sent from my iPhone
>>>> >
>>>> > On Aug 16, 2012, at 3:42 PM, "vaad.fabi at gmail.com"<
>>>> vaad.fabi at gmail.com>  wrote:
>>>> >
>>>> >>   Seems to be ok for a simple dialer, but i forgot to mention - i
>>>> need
>>>> >> advanced options for dialers, such as: IM (xmpp\facebook\twitter
>>>> >> integration), video calls, balance state, check direction rate,
>>>> >> callback, sms, top up and etc. It's smth like UC application with
>>>> >> calling support as one of the many options.
>>>> >>
>>>> >>
>>>> >> On 08/16/2012 12:22 PM, Mohammad Emran wrote:
>>>> >>> Try www.sipmobiledialer.com
>>>> >>>
>>>> >>>
>>>> >>> Sent from my iPad
>>>> >>>
>>>> >>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com"<
>>>> vaad.fabi at gmail.com>   wrote:
>>>> >>>
>>>> >>>>   Hi all,
>>>> >>>>
>>>> >>>> Please suggest mobile voip dialer vendor for
>>>> >>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>>>> >>>> don't suggest google plz
>>>> >>>>
>>>> >>>> --
>>>> >>>> Best Regards,
>>>> >>>> Vadim F.
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> _________________________________________________________________________
>>>> >>>> Professional FreeSWITCH Consulting Services:
>>>> >>>> consulting at freeswitch.org
>>>> >>>> http://www.freeswitchsolutions.com
>>>> >>>>
>>>> >>>> 
>>>> >>>> 
>>>> >>>>
>>>> >>>> Official FreeSWITCH Sites
>>>> >>>> http://www.freeswitch.org
>>>> >>>> http://wiki.freeswitch.org
>>>> >>>> http://www.cluecon.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
>>>> >>>
>>>> _________________________________________________________________________
>>>> >>> Professional FreeSWITCH Consulting Services:
>>>> >>> consulting at freeswitch.org
>>>> >>> http://www.freeswitchsolutions.com
>>>> >>>
>>>> >>> 
>>>> >>> 
>>>> >>>
>>>> >>> Official FreeSWITCH Sites
>>>> >>> http://www.freeswitch.org
>>>> >>> http://wiki.freeswitch.org
>>>> &g t;>> http://www.cluecon.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
>>>> >>
>>>> >> --
>>>> >> Best Regards,
>>>> >> Vadim F.
>>>> >>
>>>> >>
>>>> >>
>>>> _________________________________________________________________________
>>>> >> Professional FreeSWITCH Consulting Services:
>>>> >> co nsulting at freeswitch.org <consulting at freeswitch.org>
>>>> >> http://www.freeswitchsolutions.com
>>>> >>
>>>> >> 
>>>> >> 
>>>> >>
>>>> >> Official FreeSWITCH Sites
>>>> >> http://www.freeswitch.org
>>>> >> http://wiki.freeswitch.org
>>>> >> http://www.cluecon.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://lists.freeswitch.org/mailman/optio%20ns/freeswitch-users>
>>>> >> http://www.freeswitch.org
>>>> >
>>>> _________________________________________________________________________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > 
>>>> > 
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://wiki.freeswitch.org
>>>> > http://www.cluecon.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
>>>>
>>>>
>>>> --
>>>> Best Regards,
>>>> Vadim F.
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>> < br /> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>
>>>
>>>  _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com     Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com  FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.free
>>>  switch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org
>>>
>>>
>>>
>>> --  Best Regards, Vadim F.
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSU BSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> --
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>> MSN: shari_786pk at hotmail.com
>> Email: shaheryarkh at googlemail.com
>>
>> ------------------------------
>> South Africa premier free email service - webmail.co.za<http://www.webmail.co.za/>
>> <http://b.wm.co.za/click.pwm?cid=20050843&loc=N-MT&seq=502dffd1>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
-------------- next part --------------
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From mitch.capper at gmail.com  Sun Aug 19 03:08:50 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Sat, 18 Aug 2012 16:08:50 -0700
Subject: [Freeswitch-users] mobile diallers
In-Reply-To: <CAFZQphz6VSFv0EGDGchJ0AYZ+=nRh3=disz-u1WjDDsnVf4CbA@mail.gmail.com>
References: <mailman.17045.1345189578.13443.freeswitch-users@lists.freeswitch.org>
	<26f073599ffd326824ba44e1ec40fa4f@www.webmail.co.za>
	<CAMbW9oLUBzXJSZk0x21fCACiZe5VWrDLUE8hwg9isgvb2yuEGQ@mail.gmail.com>
	<CAFZQphz6VSFv0EGDGchJ0AYZ+=nRh3=disz-u1WjDDsnVf4CbA@mail.gmail.com>
Message-ID: <CALn233oKcBHSQrbF+bTK+o25s2pxzsmvDnte-xRRZFkGVGv9fg@mail.gmail.com>

Do you know what the latency is for the push networks?  On an email 30-45
second delay is next to nothing but a ringing phone call that is disastrous.

~Mitch

On Sat, Aug 18, 2012 at 2:04 PM, Muhammad Shahzad <
shaheryarkh at googlemail.com> wrote:

> Yup, that's correct and the solution on Android platform is GCM (Google
> Cloud Messaging) and on iPhone APNS (Apple Push Notification Service). Such
> that whenever there is an incoming call for the device, the server sends a
> push notification to device to intimate an incoming call, the device
> suppose to active SIP client, which immediately gets SIP Registered and as
> soon as server see client has come online it sends call to it. Then as soon
> as call finishes SIP client de-register itself from SIP server and exits.
> This considerably saves battery and data overheads.
>
> Thank you.
>
>
> On Sat, Aug 18, 2012 at 10:56 PM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
>> I've tried acrobits (slightly), SipDroid, CSipSimple and the built-in
>> android client in gingerbread. They're best avoided for incoming calls if
>> you value your battery life, enabling registration for inbound calls can
>> reduce your battery life rather drastically.
>>
>>
>> On 17 August 2012 09:24, clive engelberg <clive18 at webmail.co.za> wrote:
>>
>>> Hi
>>>
>>>
>>>
>>> Someone recommended http://www.acrobits.cz
>>>
>>>
>>>
>>> Worth a try.
>>>
>>>
>>>
>>> regards
>>>
>>> Clive
>>>
>>>
>>>
>>> On Fri, 17 Aug 2012 11:46:18 +0400
>>> freeswitch-users-request at lists.freeswitch.org wrote
>>>
>>>  IMSDroid is based on Doubango developed by Doubango Telecom, which
>>> also has support for iPhone, Windows and Mac.
>>> You should check their website at http://www.doubango.org. I think they
>>> may provide you commercial support for their productions as well, just talk
>>> to them.
>>> Thank you.
>>>
>>>
>>> On Fri, Aug 17, 2012 at 9:05 AM, vaad.fabi at gmail.com <
>>> vaad.fabi at gmail.com> wrote:
>>>
>>>>
>>>> IMSDROID - only Android platform, no  centralizied dev team\vendor
>>>> with  support and SLA
>>>> SIPDROID - only Android platform, no  centralizied dev team\vendor
>>>> with  support and SLA
>>>> Counterpath - as i know no mobile platforms
>>>>
>>>> Searching...
>>>>
>>>>
>>>>
>>>> On 08/16/2012 08:58 PM, freeswitch wrote:
>>>>
>>>>  IMSDROID will be good!
>>>>
>>>>
>>>>  At 2012-08-16 20:44:54,BookBag <asaad2 at gmail.com> <asaad2 at gmail.com>wrote:
>>>>
>>>> I'm surprised no one said Bria by counterpath. Which allows for xmpp
>>>> and presence. The softphone allows you to send sms also if you buy the full
>>>> version. Another one that I tested is Csipsimple, this one allows sms for
>>>> free. But my absolute favorite is Zoiper, unfortunately no sms capability
>>>> but wasnt buggy except for one bug I encountered with sending the correct
>>>> dtmf style. Thank god their new update took care of that. Phone now seems
>>>> very solid and now includes a QR reader.
>>>>
>>>> On Thu, Aug 16, 2012 at 6:16 AM, vaad.fabi at gmail.com <
>>>> vaad.fabi at gmail.com> wrote:
>>>>
>>>>>   Does they provide these software development, customisation and
>>>>> support ? i think no.  I need vendor\development team who can provide
>>>>> development\customisation\support.
>>>>> Anyway thx.
>>>>>
>>>>>
>>>>> On 08/16/2012 12:53 PM, Mohammad Emran wrote:
>>>>> > Then try nimbuzz or fring.... If it works for u.
>>>>> >
>>>>> > Sent from my iPhone
>>>>> >
>>>>> > On Aug 16, 2012, at 3:42 PM, "vaad.fabi at gmail.com"<
>>>>> vaad.fabi at gmail.com>  wrote:
>>>>> >
>>>>> >>   Seems to be ok for a simple dialer, but i forgot to mention - i
>>>>> need
>>>>> >> advanced options for dialers, such as: IM (xmpp\facebook\twitter
>>>>> >> integration), video calls, balance state, check direction rate,
>>>>> >> callback, sms, top up and etc. It's smth like UC application with
>>>>> >> calling support as one of the many options.
>>>>> >>
>>>>> >>
>>>>> >> On 08/16/2012 12:22 PM, Mohammad Emran wrote:
>>>>> >>> Try www.sipmobiledialer.com
>>>>> >>>
>>>>> >>>
>>>>> >>> Sent from my iPad
>>>>> >>>
>>>>> >>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com"<
>>>>> vaad.fabi at gmail.com>   wrote:
>>>>> >>>
>>>>> >>>>   Hi all,
>>>>> >>>>
>>>>> >>>> Please suggest mobile voip dialer vendor for
>>>>> >>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>>>>> >>>> don't suggest google plz
>>>>> >>>>
>>>>> >>>> --
>>>>> >>>> Best Regards,
>>>>> >>>> Vadim F.
>>>>> >>>>
>>>>> >>>>
>>>>> >>>>
>>>>> _________________________________________________________________________
>>>>> >>>> Professional FreeSWITCH Consulting Services:
>>>>> >>>> consulting at freeswitch.org
>>>>> >>>> http://www.freeswitchsolutions.com
>>>>> >>>>
>>>>> >>>> 
>>>>> >>>> 
>>>>> >>>>
>>>>> >>>> Official FreeSWITCH Sites
>>>>> >>>> http://www.freeswitch.org
>>>>> >>>> http://wiki.freeswitch.org
>>>>> >>>> http://www.cluecon.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
>>>>> >>>
>>>>> _________________________________________________________________________
>>>>> >>> Professional FreeSWITCH Consulting Services:
>>>>> >>> consulting at freeswitch.org
>>>>> >>> http://www.freeswitchsolutions.com
>>>>> >>>
>>>>> >>> 
>>>>> >>> 
>>>>> >>>
>>>>> >>> Official FreeSWITCH Sites
>>>>> >>> http://www.freeswitch.org
>>>>> >>> http://wiki.freeswitch.org
>>>>> &g t;>> http://www.cluecon.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
>>>>> >>
>>>>> >> --
>>>>> >> Best Regards,
>>>>> >> Vadim F.
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> _________________________________________________________________________
>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>> >> co nsulting at freeswitch.org <consulting at freeswitch.org>
>>>>> >> http://www.freeswitchsolutions.com
>>>>> >>
>>>>> >> 
>>>>> >> 
>>>>> >>
>>>>> >> Official FreeSWITCH Sites
>>>>> >> http://www.freeswitch.org
>>>>> >> http://wiki.freeswitch.org
>>>>> >> http://www.cluecon.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://lists.freeswitch.org/mailman/optio%20ns/freeswitch-users>
>>>>> >> http://www.freeswitch.org
>>>>> >
>>>>> _________________________________________________________________________
>>>>> > Professional FreeSWITCH Consulting Services:
>>>>> > consulting at freeswitch.org
>>>>> > http://www.freeswitchsolutions.com
>>>>> >
>>>>> > 
>>>>> > 
>>>>> >
>>>>> > Official FreeSWITCH Sites
>>>>> > http://www.freeswitch.org
>>>>> > http://wiki.freeswitch.org
>>>>> > http://www.cluecon.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
>>>>>
>>>>>
>>>>> --
>>>>> Best Regards,
>>>>> Vadim F.
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>> < br /> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>
>>>>
>>>>  _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com     Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com  FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.free
>>>>  switch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> --  Best Regards, Vadim F.
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSU BSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> --
>>> Muhammad Shahzad
>>> -----------------------------------
>>> CISCO Rich Media Communication Specialist (CRMCS)
>>> CISCO Certified Network Associate (CCNA)
>>> Cell: +92 334 422 40 88
>>> MSN: shari_786pk at hotmail.com
>>> Email: shaheryarkh at googlemail.com
>>>
>>> ------------------------------
>>> South Africa premier free email service - webmail.co.za<http://www.webmail.co.za/>
>>> <http://b.wm.co.za/click.pwm?cid=20050843&loc=N-MT&seq=502dffd1>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>>
>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>  *03300 100 961
>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lloyd.aloysius at gmail.com  Sun Aug 19 03:16:45 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Sat, 18 Aug 2012 19:16:45 -0400
Subject: [Freeswitch-users] [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] - call
	disconnect.
Message-ID: <CAGvuWYbeOttzsfGykpJ3OLQ9+SF1L0H2-U8i8MdxQr+d_Zs-CA@mail.gmail.com>

Hi All:

Today I download the new version from git. I have a trange behavior
[RECOVERY_ON_TIMER_EXPIRE]  and call drop.

All I am doing is just dial music on hold dial plan . Call Connect and
after 10~14 minutes  the call drop. I can see the following from the
freeswitch cli

2012-08-18 17:38:53.479895 [DEBUG] switch_core_session.c:918 Send signal
sofia/sipinterface_1/dave_barry at alcan.mydomain.ca[BREAK]
2012-08-18 17:38:53.479895 [DEBUG] switch_core_session.c:918 Send signal
sofia/sipinterface_1/dave_barry at alcan.mydomain.ca[BREAK]
2012-08-18 17:38:53.479895 [DEBUG] switch_core_session.c:918 Send signal
sofia/sipinterface_1/dave_barry at alcan.mydomain.ca[BREAK]
2012-08-18 17:38:53.499900 [DEBUG] sofia.c:6041 Channel
sofia/sipinterface_1/dave_barry at alcan.mydomain.caentering state
[terminated][408]
2012-08-18 17:38:53.499900 [DEBUG] switch_channel.c:2919
(sofia/sipinterface_1/dave_barry at alcan. mydomain.ca ) Callstate Change
ACTIVE -> HANGUP
*2012-08-18 17:38:53.499900 [NOTICE] sofia.c:6836 Hangup
sofia/sipinterface_1/dave_barry at alcan.mydomain.ca[CS_EXECUTE]
[RECOVERY_ON_TIMER_EXPIRE]*
2012-08-18 17:38:53.499900 [DEBUG] switch_channel.c:2942 Send signal
sofia/sipinterface_1/dave_barry at alcan.mydomain.ca[KILL]
2012-08-18 17:38:53.499900 [DEBUG] switch_core_session.c:1223 Send signal
sofia/sipinterface_1/dave_barry at alcan.mydomain.ca[BREAK]
2012-08-18 17:38:53.499900 [DEBUG] switch_ivr_play_say.c:1682 done playing
file local_stream://moh
2012-08-18 17:38:53.499900 [DEBUG] switch_core_session.c:2347
sofia/sipinterface_1/dave_barry at alcan.mydomain.caskip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2012-08-18 17:38:53.499900 [DEBUG] switch_core_state_machine.c:440
(sofia/sipinterface_1/dave_barry at alcan. mydomain.ca ) State EXECUTE going
to sleep



Is this a bug?

Thanks
Lloyd
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From shaheryarkh at googlemail.com  Sun Aug 19 03:30:17 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Sun, 19 Aug 2012 01:30:17 +0200
Subject: [Freeswitch-users] mobile diallers
In-Reply-To: <CALn233oKcBHSQrbF+bTK+o25s2pxzsmvDnte-xRRZFkGVGv9fg@mail.gmail.com>
References: <mailman.17045.1345189578.13443.freeswitch-users@lists.freeswitch.org>
	<26f073599ffd326824ba44e1ec40fa4f@www.webmail.co.za>
	<CAMbW9oLUBzXJSZk0x21fCACiZe5VWrDLUE8hwg9isgvb2yuEGQ@mail.gmail.com>
	<CAFZQphz6VSFv0EGDGchJ0AYZ+=nRh3=disz-u1WjDDsnVf4CbA@mail.gmail.com>
	<CALn233oKcBHSQrbF+bTK+o25s2pxzsmvDnte-xRRZFkGVGv9fg@mail.gmail.com>
Message-ID: <CAFZQphz0MTxxbYf7-q8aCFt50fLV9q-V6hUNYu3G2ztSAWkQWA@mail.gmail.com>

Not only I have developed such solutions but also have done extensive
testing. Per my experience APNS latency is 3 to 5 seconds (even less if you
have iPhone 4S or better) while GCM  latency is between 5 to 8 seconds
(even less if you have some high end device like Samsung Galaxy S3), which
is not so bad, you only need to keep caller busy during this time perhaps
by giving him fake ring or play some IVR in early media etc.

Try it yourself,

http://vopium.com
http://viber.com

I prefer Vopium since it has better price and quality.

Thank you.


On Sun, Aug 19, 2012 at 1:08 AM, Mitch Capper <mitch.capper at gmail.com>wrote:

> Do you know what the latency is for the push networks?  On an email 30-45
> second delay is next to nothing but a ringing phone call that is disastrous.
>
> ~Mitch
>
>
> On Sat, Aug 18, 2012 at 2:04 PM, Muhammad Shahzad <
> shaheryarkh at googlemail.com> wrote:
>
>> Yup, that's correct and the solution on Android platform is GCM (Google
>> Cloud Messaging) and on iPhone APNS (Apple Push Notification Service). Such
>> that whenever there is an incoming call for the device, the server sends a
>> push notification to device to intimate an incoming call, the device
>> suppose to active SIP client, which immediately gets SIP Registered and as
>> soon as server see client has come online it sends call to it. Then as soon
>> as call finishes SIP client de-register itself from SIP server and exits.
>> This considerably saves battery and data overheads.
>>
>> Thank you.
>>
>>
>> On Sat, Aug 18, 2012 at 10:56 PM, Andrew Cassidy <
>> andrew at cassidywebservices.co.uk> wrote:
>>
>>> I've tried acrobits (slightly), SipDroid, CSipSimple and the built-in
>>> android client in gingerbread. They're best avoided for incoming calls if
>>> you value your battery life, enabling registration for inbound calls can
>>> reduce your battery life rather drastically.
>>>
>>>
>>> On 17 August 2012 09:24, clive engelberg <clive18 at webmail.co.za> wrote:
>>>
>>>> Hi
>>>>
>>>>
>>>>
>>>> Someone recommended http://www.acrobits.cz
>>>>
>>>>
>>>>
>>>> Worth a try.
>>>>
>>>>
>>>>
>>>> regards
>>>>
>>>> Clive
>>>>
>>>>
>>>>
>>>> On Fri, 17 Aug 2012 11:46:18 +0400
>>>> freeswitch-users-request at lists.freeswitch.org wrote
>>>>
>>>>  IMSDroid is based on Doubango developed by Doubango Telecom, which
>>>> also has support for iPhone, Windows and Mac.
>>>> You should check their website at http://www.doubango.org. I think
>>>> they may provide you commercial support for their productions as well, just
>>>> talk to them.
>>>> Thank you.
>>>>
>>>>
>>>> On Fri, Aug 17, 2012 at 9:05 AM, vaad.fabi at gmail.com <
>>>> vaad.fabi at gmail.com> wrote:
>>>>
>>>>>
>>>>> IMSDROID - only Android platform, no  centralizied dev team\vendor
>>>>> with  support and SLA
>>>>> SIPDROID - only Android platform, no  centralizied dev team\vendor
>>>>> with  support and SLA
>>>>> Counterpath - as i know no mobile platforms
>>>>>
>>>>> Searching...
>>>>>
>>>>>
>>>>>
>>>>> On 08/16/2012 08:58 PM, freeswitch wrote:
>>>>>
>>>>>  IMSDROID will be good!
>>>>>
>>>>>
>>>>>  At 2012-08-16 20:44:54,BookBag <asaad2 at gmail.com> <asaad2 at gmail.com>wrote:
>>>>>
>>>>> I'm surprised no one said Bria by counterpath. Which allows for xmpp
>>>>> and presence. The softphone allows you to send sms also if you buy the full
>>>>> version. Another one that I tested is Csipsimple, this one allows sms for
>>>>> free. But my absolute favorite is Zoiper, unfortunately no sms capability
>>>>> but wasnt buggy except for one bug I encountered with sending the correct
>>>>> dtmf style. Thank god their new update took care of that. Phone now seems
>>>>> very solid and now includes a QR reader.
>>>>>
>>>>> On Thu, Aug 16, 2012 at 6:16 AM, vaad.fabi at gmail.com <
>>>>> vaad.fabi at gmail.com> wrote:
>>>>>
>>>>>>   Does they provide these software development, customisation and
>>>>>> support ? i think no.  I need vendor\development team who can provide
>>>>>> development\customisation\support.
>>>>>> Anyway thx.
>>>>>>
>>>>>>
>>>>>> On 08/16/2012 12:53 PM, Mohammad Emran wrote:
>>>>>> > Then try nimbuzz or fring.... If it works for u.
>>>>>> >
>>>>>> > Sent from my iPhone
>>>>>> >
>>>>>> > On Aug 16, 2012, at 3:42 PM, "vaad.fabi at gmail.com"<
>>>>>> vaad.fabi at gmail.com>  wrote:
>>>>>> >
>>>>>> >>   Seems to be ok for a simple dialer, but i forgot to mention - i
>>>>>> need
>>>>>> >> advanced options for dialers, such as: IM (xmpp\facebook\twitter
>>>>>> >> integration), video calls, balance state, check direction rate,
>>>>>> >> callback, sms, top up and etc. It's smth like UC application with
>>>>>> >> calling support as one of the many options.
>>>>>> >>
>>>>>> >>
>>>>>> >> On 08/16/2012 12:22 PM, Mohammad Emran wrote:
>>>>>> >>> Try www.sipmobiledialer.com
>>>>>> >>>
>>>>>> >>>
>>>>>> >>> Sent from my iPad
>>>>>> >>>
>>>>>> >>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com"<
>>>>>> vaad.fabi at gmail.com>   wrote:
>>>>>> >>>
>>>>>> >>>>   Hi all,
>>>>>> >>>>
>>>>>> >>>> Please suggest mobile voip dialer vendor for
>>>>>> >>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>>>>>> >>>> don't suggest google plz
>>>>>> >>>>
>>>>>> >>>> --
>>>>>> >>>> Best Regards,
>>>>>> >>>> Vadim F.
>>>>>> >>>>
>>>>>> >>>>
>>>>>> >>>>
>>>>>> _________________________________________________________________________
>>>>>> >>>> Professional FreeSWITCH Consulting Services:
>>>>>> >>>> consulting at freeswitch.org
>>>>>> >>>> http://www.freeswitchsolutions.com
>>>>>> >>>>
>>>>>> >>>> 
>>>>>> >>>> 
>>>>>> >>>>
>>>>>> >>>> Official FreeSWITCH Sites
>>>>>> >>>> http://www.freeswitch.org
>>>>>> >>>> http://wiki.freeswitch.org
>>>>>> >>>> http://www.cluecon.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
>>>>>> >>>
>>>>>> _________________________________________________________________________
>>>>>> >>> Professional FreeSWITCH Consulting Services:
>>>>>> >>> consulting at freeswitch.org
>>>>>> >>> http://www.freeswitchsolutions.com
>>>>>> >>>
>>>>>> >>> 
>>>>>> >>> 
>>>>>> >>>
>>>>>> >>> Official FreeSWITCH Sites
>>>>>> >>> http://www.freeswitch.org
>>>>>> >>> http://wiki.freeswitch.org
>>>>>> &g t;>> http://www.cluecon.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
>>>>>> >>
>>>>>> >> --
>>>>>> >> Best Regards,
>>>>>> >> Vadim F.
>>>>>> >>
>>>>>> >>
>>>>>> >>
>>>>>> _________________________________________________________________________
>>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>>> >> co nsulting at freeswitch.org <consulting at freeswitch.org>
>>>>>> >> http://www.freeswitchsolutions.com
>>>>>> >>
>>>>>> >> 
>>>>>> >> 
>>>>>> >>
>>>>>> >> Official FreeSWITCH Sites
>>>>>> >> http://www.freeswitch.org
>>>>>> >> http://wiki.freeswitch.org
>>>>>> >> http://www.cluecon.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://lists.freeswitch.org/mailman/optio%20ns/freeswitch-users>
>>>>>> >> http://www.freeswitch.org
>>>>>> >
>>>>>> _________________________________________________________________________
>>>>>> > Professional FreeSWITCH Consulting Services:
>>>>>> > consulting at freeswitch.org
>>>>>> > http://www.freeswitchsolutions.com
>>>>>> >
>>>>>> > 
>>>>>> > 
>>>>>> >
>>>>>> > Official FreeSWITCH Sites
>>>>>> > http://www.freeswitch.org
>>>>>> > http://wiki.freeswitch.org
>>>>>> > http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Best Regards,
>>>>>> Vadim F.
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>> < br /> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>
>>>>>
>>>>>  _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com     Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com  FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.free
>>>>>  switch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> --  Best Regards, Vadim F.
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSU BSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> --
>>>> Muhammad Shahzad
>>>> -----------------------------------
>>>> CISCO Rich Media Communication Specialist (CRMCS)
>>>> CISCO Certified Network Associate (CCNA)
>>>> Cell: +92 334 422 40 88
>>>> MSN: shari_786pk at hotmail.com
>>>> Email: shaheryarkh at googlemail.com
>>>>
>>>> ------------------------------
>>>> South Africa premier free email service - webmail.co.za<http://www.webmail.co.za/>
>>>> <http://b.wm.co.za/click.pwm?cid=20050843&loc=N-MT&seq=502dffd1>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>>
>>> --
>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>> Managing Director
>>>
>>>
>>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>>  *03300 100 961
>>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>>
>> --
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>> MSN: shari_786pk at hotmail.com
>> Email: shaheryarkh at googlemail.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From krice at freeswitch.org  Sun Aug 19 03:43:24 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 18 Aug 2012 18:43:24 -0500
Subject: [Freeswitch-users] [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] -
 call disconnect.
In-Reply-To: <CAGvuWYbeOttzsfGykpJ3OLQ9+SF1L0H2-U8i8MdxQr+d_Zs-CA@mail.gmail.com>
Message-ID: <CC5592CC.3C330%krice@freeswitch.org>

You need to get a full sip trace to see whats going on here... I suspect you
have a session or rtp timer expiring


On 8/18/12 6:16 PM, "Lloyd Aloysius" <lloyd.aloysius at gmail.com> wrote:

> Hi All:
> 
> Today I download the new version from git. I have a trange?behavior?
> [RECOVERY_ON_TIMER_EXPIRE]? and call drop.?
> 
> All I am doing is just dial music on hold?dial plan?. Call Connect and after
> 10~14 minutes ?the call drop. I can see the following from the freeswitch cli
> 
> 2012-08-18 17:38:53.479895 [DEBUG] switch_core_session.c:918 Send signal
> sofia/sipinterface_1/dave_barry at alcan.mydomain.ca[BREAK]
> 2012-08-18 17:38:53.479895 [DEBUG] switch_core_session.c:918 Send signal
> sofia/sipinterface_1/dave_barry at alcan.mydomain.ca[BREAK]
> 2012-08-18 17:38:53.479895 [DEBUG] switch_core_session.c:918 Send signal
> sofia/sipinterface_1/dave_barry at alcan.mydomain.ca[BREAK]
> 2012-08-18 17:38:53.499900 [DEBUG] sofia.c:6041 Channel
> sofia/sipinterface_1/dave_barry at alcan.mydomain.caentering state
> [terminated][408]
> 2012-08-18 17:38:53.499900 [DEBUG] switch_channel.c:2919
> (sofia/sipinterface_1/dave_barry at alcan. mydomain.ca <http://mydomain.ca> ?)
> Callstate Change ACTIVE -> HANGUP
> 2012-08-18 17:38:53.499900 [NOTICE] sofia.c:6836 Hangup
> sofia/sipinterface_1/dave_barry at alcan.mydomain.ca[CS_EXECUTE]
> [RECOVERY_ON_TIMER_EXPIRE]
> 2012-08-18 17:38:53.499900 [DEBUG] switch_channel.c:2942 Send signal
> sofia/sipinterface_1/dave_barry at alcan.mydomain.ca[KILL]
> 2012-08-18 17:38:53.499900 [DEBUG] switch_core_session.c:1223 Send signal
> sofia/sipinterface_1/dave_barry at alcan.mydomain.ca[BREAK]
> 2012-08-18 17:38:53.499900 [DEBUG] switch_ivr_play_say.c:1682 done playing
> file local_stream://moh
> 2012-08-18 17:38:53.499900 [DEBUG] switch_core_session.c:2347
> sofia/sipinterface_1/dave_barry at alcan.mydomain.caskip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2012-08-18 17:38:53.499900 [DEBUG] switch_core_state_machine.c:440
> (sofia/sipinterface_1/dave_barry at alcan. mydomain.ca <http://mydomain.ca> ?)
> State EXECUTE going to sleep
> 
> 
> 
> Is this a bug?
> 
> Thanks
> Lloyd
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From ian at unifiedarts.com  Sun Aug 19 20:41:19 2012
From: ian at unifiedarts.com (Willoughby, Ian)
Date: Sun, 19 Aug 2012 16:41:19 +0000
Subject: [Freeswitch-users] mod_spy codec issue
Message-ID: <CC568F6D.615B5%ian@unifiedarts.com>

I'm on the latest build and have a problem with garbled audio while executing userspy.  When I do a show channels, I see that my read_codec is set to L16 although I don't have that set if the global coded prefs.  Is there a way to force or deny codecs for userspy?  I have tried the following to no avail:

<action application="export" data="nolocal:absolute_codec_string=PCMU,PCMA"/>
        <action application="export" data="absolute_codec_string=PCMU,PCMA"/>
        <action application="answer"/>
        <action application="set" data="domain_name=blah.com"/>
        <action application="userspy" data="{absolute_codec_string=PCMU}$1@${domain_name} ${hash(select/${domain_name}/$1)}"/>

Ian

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From lists at kavun.ch  Sun Aug 19 21:01:34 2012
From: lists at kavun.ch (Emrah)
Date: Sun, 19 Aug 2012 13:01:34 -0400
Subject: [Freeswitch-users] Regex issue
Message-ID: <C92661B1-1476-4FBE-9181-34DEAD7AADC7@kavun.ch>

Hi all,

I am trying to e164ize the caller id number received on my inbound dids and thought I would experiment with multiple conditions first. My test is to only allow user 20 and 21 to call 311 in NYC. Note that I already have 311 working by using another evaluation process.

I know I can optimize the caller id conditions by piping each possibility, but it's not what's not working. I'd like to keep it that way for experimentation purposes.

The line that I suspect causing the issue is highlighted in my pastebin post, available here:
http://pastebin.freeswitch.org/19735

Your help would be greatly appreciated.

thanks,

--
Emrah

?In theory, theory and practice are the same. In practice, they are not.? Albert Einstein

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From lists at kavun.ch  Sun Aug 19 21:08:21 2012
From: lists at kavun.ch (Emrah)
Date: Sun, 19 Aug 2012 13:08:21 -0400
Subject: [Freeswitch-users] CALL TIMEOUT and interprise bridge
In-Reply-To: <00a201cd7bd5$9382dd50$ba8897f0$@bizfocused.com>
References: <00a201cd7bd5$9382dd50$ba8897f0$@bizfocused.com>
Message-ID: <4B032411-ABF3-400A-9C77-5768CF5E599B@kavun.ch>

Use <originate_timeout=25> before your dial string, you don't need to specify it per leg.

Hope this helps,
--
Emrah

?In theory, theory and practice are the same. In practice, they are not.? Albert Einstein

On Aug 16, 2012, at 1:35 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Hi,
> I have noticed that ?SET [call_timeout]=[25]? seems to be ignored when using an Enterprise bridge (:_:) to two extensions.  The timeout is always 60 seconds.
>  
> Do I just need to set the timeout in {} on each leg?
>  
> Sean
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian.wiese.freeswitch at gmail.com  Sun Aug 19 21:14:53 2012
From: brian.wiese.freeswitch at gmail.com (Brian Wiese)
Date: Sun, 19 Aug 2012 12:14:53 -0500
Subject: [Freeswitch-users] Regex issue
In-Reply-To: <C92661B1-1476-4FBE-9181-34DEAD7AADC7@kavun.ch>
References: <C92661B1-1476-4FBE-9181-34DEAD7AADC7@kavun.ch>
Message-ID: <CADEQS5y=-Ur5WYeBR5TW=BC2-K447KtpgfjfE__OU9NECQc9Fg@mail.gmail.com>

Emrah:

Try <regex field="${allow_311}" expression="^true$" />

If that doesn't work please pastebin a log of a call.

~Brian

On Sun, Aug 19, 2012 at 12:01 PM, Emrah <lists at kavun.ch> wrote:
> Hi all,
>
> I am trying to e164ize the caller id number received on my inbound dids and
> thought I would experiment with multiple conditions first. My test is to
> only allow user 20 and 21 to call 311 in NYC. Note that I already have 311
> working by using another evaluation process.
>
> I know I can optimize the caller id conditions by piping each possibility,
> but it's not what's not working. I'd like to keep it that way for
> experimentation purposes.
>
> The line that I suspect causing the issue is highlighted in my pastebin
> post, available here:
> http://pastebin.freeswitch.org/19735
>
> Your help would be greatly appreciated.
>
> thanks,
>
> --
> Emrah
>
> ?In theory, theory and practice are the same. In practice, they are not.?
> Albert Einstein
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 vbvbrj at gmail.com  Mon Aug 20 10:29:29 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Mon, 20 Aug 2012 09:29:29 +0300
Subject: [Freeswitch-users] Recording problem
In-Reply-To: <CAAdTMa8DdeRCdYGRKeD2qJLAZyfLr9Sv08-R=_YMNSaSH_svVQ@mail.gmail.com>
References: <502E7201.5050909@gmail.com> <502E748C.7000900@gmail.com>
	<CAAdTMa_n-rQay6e+OUrB1w9qYS3s017Nj1SJTBYOCB4xo9wiBg@mail.gmail.com>
	<502E7F06.8090307@gmail.com>
	<CAAdTMa8DdeRCdYGRKeD2qJLAZyfLr9Sv08-R=_YMNSaSH_svVQ@mail.gmail.com>
Message-ID: <5031D949.7040100@gmail.com>

In my DPH-150S for the problematic user in config at web management I 
switched "Signal Standard" from China to Russian, and this seems to 
solve the record problem. Why this is so influencing - I don't understand.
How I will resolve the same issue for PAP2T?


From vbvbrj at gmail.com  Mon Aug 20 10:31:40 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Mon, 20 Aug 2012 09:31:40 +0300
Subject: [Freeswitch-users] mod_callcenter record template
Message-ID: <5031D9CC.7060300@gmail.com>

In mod callcenter.conf.xml there is:
<param name="record-template" 
value="$${base_dir}/recordings/sales/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>
This names the file with callers number and callcenter's assigned 
number. Bit how to include in the file name the agent's number, which 
responded? Which variable to use?


From sameer2k3t at gmail.com  Mon Aug 20 11:29:57 2012
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Mon, 20 Aug 2012 12:29:57 +0500
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
Message-ID: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>

Hi,

I can't get audio working with e1550

I have this config :
    <interface id="1" name="gsm01">
      <param name="controldevice_name" value="/dev/ttyUSB0"/>
      <param name="controldevice_audio_name" value="/dev/ttyUSB3"/>

I tried swaping 0 and 3 but having same results

I am originating a call from softphone and answering it on cell phone but
softphone doesn't get notified of being answered and keeps ringing

Please help

Thanks
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From andrew at cassidywebservices.co.uk  Mon Aug 20 11:45:37 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Mon, 20 Aug 2012 08:45:37 +0100
Subject: [Freeswitch-users] mobile diallers
In-Reply-To: <CAFZQphz0MTxxbYf7-q8aCFt50fLV9q-V6hUNYu3G2ztSAWkQWA@mail.gmail.com>
References: <mailman.17045.1345189578.13443.freeswitch-users@lists.freeswitch.org>
	<26f073599ffd326824ba44e1ec40fa4f@www.webmail.co.za>
	<CAMbW9oLUBzXJSZk0x21fCACiZe5VWrDLUE8hwg9isgvb2yuEGQ@mail.gmail.com>
	<CAFZQphz6VSFv0EGDGchJ0AYZ+=nRh3=disz-u1WjDDsnVf4CbA@mail.gmail.com>
	<CALn233oKcBHSQrbF+bTK+o25s2pxzsmvDnte-xRRZFkGVGv9fg@mail.gmail.com>
	<CAFZQphz0MTxxbYf7-q8aCFt50fLV9q-V6hUNYu3G2ztSAWkQWA@mail.gmail.com>
Message-ID: <CAMbW9oJj7rOkyZDagvyrTXR4hV_2bpCEiCG7UTtstN+bO1UEVg@mail.gmail.com>

In the UK we also have solutions like this:
http://aaisp.com/kb-telecoms-sip.html

It's a mobile SIM card that you associate with a SIP account, so you do
your SIP calls technically as normal mobile calls.

You do have to pay for both incoming and outgoing calls to the device
though, but when you have no 3G/Data coverage you can still make
and receive calls and it doesn't kill your battery life.

On 19 August 2012 00:30, Muhammad Shahzad <shaheryarkh at googlemail.com>wrote:

> Not only I have developed such solutions but also have done extensive
> testing. Per my experience APNS latency is 3 to 5 seconds (even less if you
> have iPhone 4S or better) while GCM  latency is between 5 to 8 seconds
> (even less if you have some high end device like Samsung Galaxy S3), which
> is not so bad, you only need to keep caller busy during this time perhaps
> by giving him fake ring or play some IVR in early media etc.
>
> Try it yourself,
>
> http://vopium.com
> http://viber.com
>
> I prefer Vopium since it has better price and quality.
>
> Thank you.
>
>
> On Sun, Aug 19, 2012 at 1:08 AM, Mitch Capper <mitch.capper at gmail.com>wrote:
>
>> Do you know what the latency is for the push networks?  On an email 30-45
>> second delay is next to nothing but a ringing phone call that is disastrous.
>>
>> ~Mitch
>>
>>
>> On Sat, Aug 18, 2012 at 2:04 PM, Muhammad Shahzad <
>> shaheryarkh at googlemail.com> wrote:
>>
>>> Yup, that's correct and the solution on Android platform is GCM (Google
>>> Cloud Messaging) and on iPhone APNS (Apple Push Notification Service). Such
>>> that whenever there is an incoming call for the device, the server sends a
>>> push notification to device to intimate an incoming call, the device
>>> suppose to active SIP client, which immediately gets SIP Registered and as
>>> soon as server see client has come online it sends call to it. Then as soon
>>> as call finishes SIP client de-register itself from SIP server and exits.
>>> This considerably saves battery and data overheads.
>>>
>>> Thank you.
>>>
>>>
>>> On Sat, Aug 18, 2012 at 10:56 PM, Andrew Cassidy <
>>> andrew at cassidywebservices.co.uk> wrote:
>>>
>>>> I've tried acrobits (slightly), SipDroid, CSipSimple and the built-in
>>>> android client in gingerbread. They're best avoided for incoming calls if
>>>> you value your battery life, enabling registration for inbound calls can
>>>> reduce your battery life rather drastically.
>>>>
>>>>
>>>> On 17 August 2012 09:24, clive engelberg <clive18 at webmail.co.za> wrote:
>>>>
>>>>> Hi
>>>>>
>>>>>
>>>>>
>>>>> Someone recommended http://www.acrobits.cz
>>>>>
>>>>>
>>>>>
>>>>> Worth a try.
>>>>>
>>>>>
>>>>>
>>>>> regards
>>>>>
>>>>> Clive
>>>>>
>>>>>
>>>>>
>>>>> On Fri, 17 Aug 2012 11:46:18 +0400
>>>>> freeswitch-users-request at lists.freeswitch.org wrote
>>>>>
>>>>>  IMSDroid is based on Doubango developed by Doubango Telecom, which
>>>>> also has support for iPhone, Windows and Mac.
>>>>> You should check their website at http://www.doubango.org. I think
>>>>> they may provide you commercial support for their productions as well, just
>>>>> talk to them.
>>>>> Thank you.
>>>>>
>>>>>
>>>>> On Fri, Aug 17, 2012 at 9:05 AM, vaad.fabi at gmail.com <
>>>>> vaad.fabi at gmail.com> wrote:
>>>>>
>>>>>>
>>>>>> IMSDROID - only Android platform, no  centralizied dev team\vendor
>>>>>> with  support and SLA
>>>>>> SIPDROID - only Android platform, no  centralizied dev team\vendor
>>>>>> with  support and SLA
>>>>>> Counterpath - as i know no mobile platforms
>>>>>>
>>>>>> Searching...
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 08/16/2012 08:58 PM, freeswitch wrote:
>>>>>>
>>>>>>  IMSDROID will be good!
>>>>>>
>>>>>>
>>>>>>  At 2012-08-16 20:44:54,BookBag <asaad2 at gmail.com> <asaad2 at gmail.com>wrote:
>>>>>>
>>>>>> I'm surprised no one said Bria by counterpath. Which allows for xmpp
>>>>>> and presence. The softphone allows you to send sms also if you buy the full
>>>>>> version. Another one that I tested is Csipsimple, this one allows sms for
>>>>>> free. But my absolute favorite is Zoiper, unfortunately no sms capability
>>>>>> but wasnt buggy except for one bug I encountered with sending the correct
>>>>>> dtmf style. Thank god their new update took care of that. Phone now seems
>>>>>> very solid and now includes a QR reader.
>>>>>>
>>>>>> On Thu, Aug 16, 2012 at 6:16 AM, vaad.fabi at gmail.com <
>>>>>> vaad.fabi at gmail.com> wrote:
>>>>>>
>>>>>>>   Does they provide these software development, customisation and
>>>>>>> support ? i think no.  I need vendor\development team who can provide
>>>>>>> development\customisation\support.
>>>>>>> Anyway thx.
>>>>>>>
>>>>>>>
>>>>>>> On 08/16/2012 12:53 PM, Mohammad Emran wrote:
>>>>>>> > Then try nimbuzz or fring.... If it works for u.
>>>>>>> >
>>>>>>> > Sent from my iPhone
>>>>>>> >
>>>>>>> > On Aug 16, 2012, at 3:42 PM, "vaad.fabi at gmail.com"<
>>>>>>> vaad.fabi at gmail.com>  wrote:
>>>>>>> >
>>>>>>> >>   Seems to be ok for a simple dialer, but i forgot to mention - i
>>>>>>> need
>>>>>>> >> advanced options for dialers, such as: IM (xmpp\facebook\twitter
>>>>>>> >> integration), video calls, balance state, check direction rate,
>>>>>>> >> callback, sms, top up and etc. It's smth like UC application with
>>>>>>> >> calling support as one of the many options.
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> On 08/16/2012 12:22 PM, Mohammad Emran wrote:
>>>>>>> >>> Try www.sipmobiledialer.com
>>>>>>> >>>
>>>>>>> >>>
>>>>>>> >>> Sent from my iPad
>>>>>>> >>>
>>>>>>> >>> On Aug 16, 2012, at 3:15 PM, "vaad.fabi at gmail.com"<
>>>>>>> vaad.fabi at gmail.com>   wrote:
>>>>>>> >>>
>>>>>>> >>>>   Hi all,
>>>>>>> >>>>
>>>>>>> >>>> Please suggest mobile voip dialer vendor for
>>>>>>> >>>> iPhone,Android,Symbian,Blackberry and WinMobile platforms.
>>>>>>> >>>> don't suggest google plz
>>>>>>> >>>>
>>>>>>> >>>> --
>>>>>>> >>>> Best Regards,
>>>>>>> >>>> Vadim F.
>>>>>>> >>>>
>>>>>>> >>>>
>>>>>>> >>>>
>>>>>>> _________________________________________________________________________
>>>>>>> >>>> Professional FreeSWITCH Consulting Services:
>>>>>>> >>>> consulting at freeswitch.org
>>>>>>> >>>> http://www.freeswitchsolutions.com
>>>>>>> >>>>
>>>>>>> >>>> 
>>>>>>> >>>> 
>>>>>>> >>>>
>>>>>>> >>>> Official FreeSWITCH Sites
>>>>>>> >>>> http://www.freeswitch.org
>>>>>>> >>>> http://wiki.freeswitch.org
>>>>>>> >>>> http://www.cluecon.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
>>>>>>> >>>
>>>>>>> _________________________________________________________________________
>>>>>>> >>> Professional FreeSWITCH Consulting Services:
>>>>>>> >>> consulting at freeswitch.org
>>>>>>> >>> http://www.freeswitchsolutions.com
>>>>>>> >>>
>>>>>>> >>> 
>>>>>>> >>> 
>>>>>>> >>>
>>>>>>> >>> Official FreeSWITCH Sites
>>>>>>> >>> http://www.freeswitch.org
>>>>>>> >>> http://wiki.freeswitch.org
>>>>>>> &g t;>> http://www.cluecon.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
>>>>>>> >>
>>>>>>> >> --
>>>>>>> >> Best Regards,
>>>>>>> >> Vadim F.
>>>>>>> >>
>>>>>>> >>
>>>>>>> >>
>>>>>>> _________________________________________________________________________
>>>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>>>> >> co nsulting at freeswitch.org <consulting at freeswitch.org>
>>>>>>> >> http://www.freeswitchsolutions.com
>>>>>>> >>
>>>>>>> >> 
>>>>>>> >> 
>>>>>>> >>
>>>>>>> >> Official FreeSWITCH Sites
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >> http://wiki.freeswitch.org
>>>>>>> >> http://www.cluecon.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://lists.freeswitch.org/mailman/optio%20ns/freeswitch-users>
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >
>>>>>>> _________________________________________________________________________
>>>>>>> > Professional FreeSWITCH Consulting Services:
>>>>>>> > consulting at freeswitch.org
>>>>>>> > http://www.freeswitchsolutions.com
>>>>>>> >
>>>>>>> > 
>>>>>>> > 
>>>>>>> >
>>>>>>> > Official FreeSWITCH Sites
>>>>>>> > http://www.freeswitch.org
>>>>>>> > http://wiki.freeswitch.org
>>>>>>> > http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Best Regards,
>>>>>>> Vadim F.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>> < br /> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>
>>>>>>
>>>>>>  _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com     Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com  FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.free
>>>>>>  switch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>> --  Best Regards, Vadim F.
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSU BSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Muhammad Shahzad
>>>>> -----------------------------------
>>>>> CISCO Rich Media Communication Specialist (CRMCS)
>>>>> CISCO Certified Network Associate (CCNA)
>>>>> Cell: +92 334 422 40 88
>>>>> MSN: shari_786pk at hotmail.com
>>>>> Email: shaheryarkh at googlemail.com
>>>>>
>>>>> ------------------------------
>>>>> South Africa premier free email service - webmail.co.za<http://www.webmail.co.za/>
>>>>> <http://b.wm.co.za/click.pwm?cid=20050843&loc=N-MT&seq=502dffd1>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>>> Managing Director
>>>>
>>>>
>>>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>>>  *03300 100 961
>>>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>>>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>>
>>> --
>>> Muhammad Shahzad
>>> -----------------------------------
>>> CISCO Rich Media Communication Specialist (CRMCS)
>>> CISCO Certified Network Associate (CCNA)
>>> Cell: +92 334 422 40 88
>>> MSN: shari_786pk at hotmail.com
>>> Email: shaheryarkh at googlemail.com
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From dvl36.ripe.nick at gmail.com  Mon Aug 20 12:43:51 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Mon, 20 Aug 2012 11:43:51 +0300
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
Message-ID: <CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>

Hello.
Try:
 <param name="controldevice_name" value="/dev/ttyUSB2"/>
 <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
It's worked for me.

2012/8/20 Sameer Khan <sameer2k3t at gmail.com>

> Hi,
>
> I can't get audio working with e1550
>
> I have this config :
>     <interface id="1" name="gsm01">
>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>       <param name="controldevice_audio_name" value="/dev/ttyUSB3"/>
>
> I tried swaping 0 and 3 but having same results
>
> I am originating a call from softphone and answering it on cell phone but
> softphone doesn't get notified of being answered and keeps ringing
>
> Please help
>
> Thanks
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 odermann at googlemail.com  Mon Aug 20 14:07:30 2012
From: odermann at googlemail.com (Dennis)
Date: Mon, 20 Aug 2012 12:07:30 +0200
Subject: [Freeswitch-users] How to config FS to listen to early media at
	180?
In-Reply-To: <BLU0-SMTP89FB512A88818BB97CCC678CB60@phx.gbl>
References: <BLU0-SMTP89FB512A88818BB97CCC678CB60@phx.gbl>
Message-ID: <CALaOL+jMfVAODPMNgsP7nCVgntXmF08LAVvrhKzK7auy89s5cQ@mail.gmail.com>

as already written in
http://freeswitch-users.2379917.n2.nabble.com/Early-media-Sending-180-instead-of-183-td7581480.html,
we have exactly the same problem.

this is not a problem of carriers or something we could fix with them
- it's just the way it is done in germany most of the time (and other
places).

we do not say, that fs is doing it in a wrong way! it just leads to
problems for others and us, which could be avoided quite easily with
some kind of option, setting or parameter.


kind regards
dennis


From g.d.monnezza at tiscali.it  Mon Aug 20 14:53:05 2012
From: g.d.monnezza at tiscali.it (g)
Date: Mon, 20 Aug 2012 12:53:05 +0200
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <502E4B94.1040503@hw.ac.uk>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<10440306.PhsNeVk7eG@virtex> <502E4B94.1040503@hw.ac.uk>
Message-ID: <7518514.pVGd9TpNrB@virtex>

You can log an agent into a queue by a web application in which a button click 
handles commands via the FS socket, or just let the agent dial an extension 
(i.e. 999+itssoftphonenumber) where dialplan sets that agent in the correct 
state.
ie (replace agent_login regex into something like 999201 if your agent is 201)
<extension name="agent_login">
  <condition field="destination_number" expression="^agent-login$">
    <action application="set" data="res=${callcenter_config(agent set status 
${caller_id_number}@${domain_name} 'Available')}" />
    <action application="answer" data=""/>
    <action application="sleep" data="500"/>
    <action application="playback" data="ivr/ivr-you_are_now_logged_in.wav"/>
    <action application="hangup" data=""/>
  </condition>
</extension>

<extension name="agent_logoff">
  <condition field="destination_number" expression="^agent-logoff$">
    <action application="set" data="res=${callcenter_config(agent set status 
${caller_id_number}@${domain_name} 'Logged Out')}" />
    <action application="answer" data=""/>
    <action application="sleep" data="500"/>
    <action application="playback" data="ivr/ivr-you_are_now_logged_out.wav"/>
    <action application="hangup" data=""/>
  </condition>
</extension>

The important thing is tha the agent is, somehow, putted in "available" 
status. You can also do it via fs_cli, but now I can't remeber the exact 
syntax for that. Help from cli should clarify. 

At this point, the agent should result in "waiting" state
Of course, you should before declared that agent as member of a tier or, at 
least, of a queue.

Hope it helps
g

On Friday 17 August 2012 14:48:04 x.liu wrote:
> I am also trying the mod_callcenter. What I want is to route a call to
> one agent ( a registered FS client, e.g. a SIP softphone)
> of a list of agents by a Random or RoundRobin policy.
> 
> When I dial the extension which calls the callcenter app, I only have
> hold-on music. It does not route a call to a agent.
> You said " You don't need agent to dial anything to get the call from
> queue. ",
> I am wondering how I could let an agent log into the queue?
> 
> Thanks!
> 
> Xing
> 
> On 08/17/2012 01:56 PM, g wrote:
> > Me too, I'm often dealing with callcenters (inbound+outbound). I found
> > in
> > Freeswitch the easyest solution to manage queue and agents. I'm not a
> > programmer, but I have experience with Asterisk, Freeswitch and the
> > whole
> > workflow of complex callcenters.
> > My suggestion too is to use mod_callcenter. Easy, fast, powerful. You
> > don't need agent to dial anything to get the call from queue.
> > In my opinion, you can use that module or, for more complex things (ie.
> > outbound predictive) you can use the socket manager to perform  call
> > placement, call transfer, conference with thirdy-part recording  IVR
> > etc.
> > I feel the best solution, also if it will require some effort for a Java
> > programmer, is to do all that in php, which is light and fast also for
> > older PC, able to run also with thin-client low memory low-cpu
> > machines. If you can plan to turn to php and you don't have too short
> > time, I can consider to help for the Freeswitch part.
> > giuliano
> > 
> > On Thursday 16 August 2012 17:10:33 Nitin Tomer wrote:
> >> Hi,
> >> 
> >> 
> >> 
> >> I am working on a Contact Center solution. It will support mail, chat
> >> and call queries.
> >> 
> >> The requirements are:
> >> 
> >> 1. An end-customer calls, the call is handled by FreeSWITCH Auto
> >> Attendant. 2. Customer is presented with a menu and makes selection.
> >> His call is put on hold and an entry is made in my system's database
> >> for incoming queries. 3. These queries are shown to agents handling
> >> calls.
> >> 4. And agent clicks on a query, he is shown an extension where call is
> >> parked. He dials that and is connected to the customer.
> >> 
> >> 5. He talks to the customer and resolve his queries.
> >> 
> >> Please guide me on how can do it. The application will be written in
> >> Java.
> >> 
> >> I am an experienced programmers in Java/J2EE; but doesn't have much
> >> knowledge of VoIP/FreeSWITCH. I've configured FreeSWITCH on Ubuntu
> >> 12.04; and have installed X-Lite Softphones on two Windows machines.
> >> I configured these phones to work with FreeSWITCH and they are
> >> working fine.
> >> 
> >> 
> >> 
> >> I think that I'd have to use Valet Parking to park customer's call on
> >> an
> >> extension, then pass that extension to agent who will dial it and will
> >> be connected to the customer.
> >> 
> >> 
> >> 
> >> Please tell me whether this approach will work? And how should I go
> >> about it.
> >> 
> >> 
> >> 
> >> All help would be much appreciated.
> >> 
> >> 
> >> 
> >> Regards
> >> 
> >> 
> >> 
> >> Nitin
> > 
> > ________________________________________________________________________
> > _
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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 neilp at cs.stanford.edu  Mon Aug 20 15:52:59 2012
From: neilp at cs.stanford.edu (Neil Patel)
Date: Mon, 20 Aug 2012 17:22:59 +0530
Subject: [Freeswitch-users] Phantom tone with session:read()
Message-ID: <CAGy1GmOSoLOBg=T3HEjTDb9LL6nyt6SDh5SC900DmX=wpNzkMg@mail.gmail.com>

I'm playing a wav file via lua app and there is a phantom tone that plays
at a certain part of the file. It's not there when I play the file on my PC
natively (Audacity, Quicktime, etc.), but it's there when I play it with
session:read(). Any suggestions on how to debug this?
Thanks,
Neil
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From x.liu at hw.ac.uk  Mon Aug 20 16:00:14 2012
From: x.liu at hw.ac.uk (x.liu)
Date: Mon, 20 Aug 2012 13:00:14 +0100
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <7518514.pVGd9TpNrB@virtex>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<10440306.PhsNeVk7eG@virtex> <502E4B94.1040503@hw.ac.uk>
	<7518514.pVGd9TpNrB@virtex>
Message-ID: <503226CE.7040307@hw.ac.uk>

Hi g,

Thanks for your response!

My situation is that I don't want the agent to dial the extension to be 
able to accept calls.
The agents (e.g. a FS softphone client, or a Java application) just need 
to register to FS as SIP clients.

And I've not figured out what commands I should send if I use extension 
"^agent-login$" and FS socket.

So I guess I may be better not to use mod_callcenter for routing calls 
to different extensions.
just to use FS ESL and my own logics to bridge the calls.

I was curious about the reply "You don't need agent to dial anything to 
get the call from queue... "
and I thought my question is relevant to the thread, so I asked 
questions in this thread.
Apologies for that. I could have opened a different thread for the 
questions :-)

Cheers,
Xing


On 08/20/2012 11:53 AM, g wrote:
> You can log an agent into a queue by a web application in which a button click
> handles commands via the FS socket, or just let the agent dial an extension
> (i.e. 999+itssoftphonenumber) where dialplan sets that agent in the correct
> state.
> ie (replace agent_login regex into something like 999201 if your agent is 201)
> <extension name="agent_login">
>    <condition field="destination_number" expression="^agent-login$">
>      <action application="set" data="res=${callcenter_config(agent set status
> ${caller_id_number}@${domain_name} 'Available')}" />
>      <action application="answer" data=""/>
>      <action application="sleep" data="500"/>
>      <action application="playback" data="ivr/ivr-you_are_now_logged_in.wav"/>
>      <action application="hangup" data=""/>
>    </condition>
> </extension>
>
> <extension name="agent_logoff">
>    <condition field="destination_number" expression="^agent-logoff$">
>      <action application="set" data="res=${callcenter_config(agent set status
> ${caller_id_number}@${domain_name} 'Logged Out')}" />
>      <action application="answer" data=""/>
>      <action application="sleep" data="500"/>
>      <action application="playback" data="ivr/ivr-you_are_now_logged_out.wav"/>
>      <action application="hangup" data=""/>
>    </condition>
> </extension>
>
> The important thing is tha the agent is, somehow, putted in "available"
> status. You can also do it via fs_cli, but now I can't remeber the exact
> syntax for that. Help from cli should clarify.
>
> At this point, the agent should result in "waiting" state
> Of course, you should before declared that agent as member of a tier or, at
> least, of a queue.
>
> Hope it helps
> g
>
> On Friday 17 August 2012 14:48:04 x.liu wrote:
>> I am also trying the mod_callcenter. What I want is to route a call to
>> one agent ( a registered FS client, e.g. a SIP softphone)
>> of a list of agents by a Random or RoundRobin policy.
>>
>> When I dial the extension which calls the callcenter app, I only have
>> hold-on music. It does not route a call to a agent.
>> You said " You don't need agent to dial anything to get the call from
>> queue. ",
>> I am wondering how I could let an agent log into the queue?
>>
>> Thanks!
>>
>> Xing
>>
>> On 08/17/2012 01:56 PM, g wrote:
>>> Me too, I'm often dealing with callcenters (inbound+outbound). I found
>>> in
>>> Freeswitch the easyest solution to manage queue and agents. I'm not a
>>> programmer, but I have experience with Asterisk, Freeswitch and the
>>> whole
>>> workflow of complex callcenters.
>>> My suggestion too is to use mod_callcenter. Easy, fast, powerful. You
>>> don't need agent to dial anything to get the call from queue.
>>> In my opinion, you can use that module or, for more complex things (ie.
>>> outbound predictive) you can use the socket manager to perform  call
>>> placement, call transfer, conference with thirdy-part recording  IVR
>>> etc.
>>> I feel the best solution, also if it will require some effort for a Java
>>> programmer, is to do all that in php, which is light and fast also for
>>> older PC, able to run also with thin-client low memory low-cpu
>>> machines. If you can plan to turn to php and you don't have too short
>>> time, I can consider to help for the Freeswitch part.
>>> giuliano
>>>
>>> On Thursday 16 August 2012 17:10:33 Nitin Tomer wrote:
>>>> Hi,
>>>>
>>>>
>>>>
>>>> I am working on a Contact Center solution. It will support mail, chat
>>>> and call queries.
>>>>
>>>> The requirements are:
>>>>
>>>> 1. An end-customer calls, the call is handled by FreeSWITCH Auto
>>>> Attendant. 2. Customer is presented with a menu and makes selection.
>>>> His call is put on hold and an entry is made in my system's database
>>>> for incoming queries. 3. These queries are shown to agents handling
>>>> calls.
>>>> 4. And agent clicks on a query, he is shown an extension where call is
>>>> parked. He dials that and is connected to the customer.
>>>>
>>>> 5. He talks to the customer and resolve his queries.
>>>>
>>>> Please guide me on how can do it. The application will be written in
>>>> Java.
>>>>
>>>> I am an experienced programmers in Java/J2EE; but doesn't have much
>>>> knowledge of VoIP/FreeSWITCH. I've configured FreeSWITCH on Ubuntu
>>>> 12.04; and have installed X-Lite Softphones on two Windows machines.
>>>> I configured these phones to work with FreeSWITCH and they are
>>>> working fine.
>>>>
>>>>
>>>>
>>>> I think that I'd have to use Valet Parking to park customer's call on
>>>> an
>>>> extension, then pass that extension to agent who will dial it and will
>>>> be connected to the customer.
>>>>
>>>>
>>>>
>>>> Please tell me whether this approach will work? And how should I go
>>>> about it.
>>>>
>>>>
>>>>
>>>> All help would be much appreciated.
>>>>
>>>>
>>>>
>>>> Regards
>>>>
>>>>
>>>>
>>>> Nitin
>>> ________________________________________________________________________
>>> _
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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



-- 
Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012

We invite research leaders and ambitious early career researchers to 
join us in leading and driving research in key inter-disciplinary themes. 
Please see www.hw.ac.uk/researchleaders for further information and how
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Heriot-Watt University is a Scottish charity
registered under charity number SC000278.



From dgarcia at anew.com.ve  Mon Aug 20 17:36:42 2012
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Mon, 20 Aug 2012 09:06:42 -0430
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <503226CE.7040307@hw.ac.uk>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<10440306.PhsNeVk7eG@virtex> <502E4B94.1040503@hw.ac.uk>
	<7518514.pVGd9TpNrB@virtex> <503226CE.7040307@hw.ac.uk>
Message-ID: <50323D6A.3020708@anew.com.ve>

mmm,

You think that the agent is dialing an extension. It is not what is 
happening. The agent is dialing a number that invoke a mod_callcenter 
function: login/logout/not ready. As FS is like a pbx (and more than a 
pbx), it seems like an extension.

Just, registrer to FS is not enough. What happen when the agent need to 
go the restroom, take coffee, couching, or take a break? Then, you will 
need to reproduce in the agent gui the funcion of login, logout and 
break... but again, what happen if your agent gui fail? your callcenter 
service will stop because your gui fail? Also, if you just use register 
signaling, you could face a wide sort of operational issues like, what 
happen if the agent left the position and left the their seats: then 
calls will be routed to empty seats!

If you want to get the agent login/logout without the agent dial 
nothing... if I am not wrong, mod_callcenter offer functions to log and 
logout agent from fs console, so I think you can invoke them using 
execute() from lua or your script.


On 8/20/2012 7:30 AM, x.liu wrote:
> Hi g,
>
> Thanks for your response!
>
> My situation is that I don't want the agent to dial the extension to be
> able to accept calls.
> The agents (e.g. a FS softphone client, or a Java application) just need
> to register to FS as SIP clients.
>
> And I've not figured out what commands I should send if I use extension
> "^agent-login$" and FS socket.
>
> So I guess I may be better not to use mod_callcenter for routing calls
> to different extensions.
> just to use FS ESL and my own logics to bridge the calls.
>
> I was curious about the reply "You don't need agent to dial anything to
> get the call from queue... "
> and I thought my question is relevant to the thread, so I asked
> questions in this thread.
> Apologies for that. I could have opened a different thread for the
> questions :-)
>
> Cheers,
> Xing
>
>
> On 08/20/2012 11:53 AM, g wrote:
>> You can log an agent into a queue by a web application in which a button click
>> handles commands via the FS socket, or just let the agent dial an extension
>> (i.e. 999+itssoftphonenumber) where dialplan sets that agent in the correct
>> state.
>> ie (replace agent_login regex into something like 999201 if your agent is 201)
>> <extension name="agent_login">
>>     <condition field="destination_number" expression="^agent-login$">
>>       <action application="set" data="res=${callcenter_config(agent set status
>> ${caller_id_number}@${domain_name} 'Available')}" />
>>       <action application="answer" data=""/>
>>       <action application="sleep" data="500"/>
>>       <action application="playback" data="ivr/ivr-you_are_now_logged_in.wav"/>
>>       <action application="hangup" data=""/>
>>     </condition>
>> </extension>
>>
>> <extension name="agent_logoff">
>>     <condition field="destination_number" expression="^agent-logoff$">
>>       <action application="set" data="res=${callcenter_config(agent set status
>> ${caller_id_number}@${domain_name} 'Logged Out')}" />
>>       <action application="answer" data=""/>
>>       <action application="sleep" data="500"/>
>>       <action application="playback" data="ivr/ivr-you_are_now_logged_out.wav"/>
>>       <action application="hangup" data=""/>
>>     </condition>
>> </extension>
>>
>> The important thing is tha the agent is, somehow, putted in "available"
>> status. You can also do it via fs_cli, but now I can't remeber the exact
>> syntax for that. Help from cli should clarify.
>>
>> At this point, the agent should result in "waiting" state
>> Of course, you should before declared that agent as member of a tier or, at
>> least, of a queue.
>>
>> Hope it helps
>> g
>>
>> On Friday 17 August 2012 14:48:04 x.liu wrote:
>>> I am also trying the mod_callcenter. What I want is to route a call to
>>> one agent ( a registered FS client, e.g. a SIP softphone)
>>> of a list of agents by a Random or RoundRobin policy.
>>>
>>> When I dial the extension which calls the callcenter app, I only have
>>> hold-on music. It does not route a call to a agent.
>>> You said " You don't need agent to dial anything to get the call from
>>> queue. ",
>>> I am wondering how I could let an agent log into the queue?
>>>
>>> Thanks!
>>>
>>> Xing
>>>
>>> On 08/17/2012 01:56 PM, g wrote:
>>>> Me too, I'm often dealing with callcenters (inbound+outbound). I found
>>>> in
>>>> Freeswitch the easyest solution to manage queue and agents. I'm not a
>>>> programmer, but I have experience with Asterisk, Freeswitch and the
>>>> whole
>>>> workflow of complex callcenters.
>>>> My suggestion too is to use mod_callcenter. Easy, fast, powerful. You
>>>> don't need agent to dial anything to get the call from queue.
>>>> In my opinion, you can use that module or, for more complex things (ie.
>>>> outbound predictive) you can use the socket manager to perform  call
>>>> placement, call transfer, conference with thirdy-part recording  IVR
>>>> etc.
>>>> I feel the best solution, also if it will require some effort for a Java
>>>> programmer, is to do all that in php, which is light and fast also for
>>>> older PC, able to run also with thin-client low memory low-cpu
>>>> machines. If you can plan to turn to php and you don't have too short
>>>> time, I can consider to help for the Freeswitch part.
>>>> giuliano
>>>>
>>>> On Thursday 16 August 2012 17:10:33 Nitin Tomer wrote:
>>>>> Hi,
>>>>>
>>>>>
>>>>>
>>>>> I am working on a Contact Center solution. It will support mail, chat
>>>>> and call queries.
>>>>>
>>>>> The requirements are:
>>>>>
>>>>> 1. An end-customer calls, the call is handled by FreeSWITCH Auto
>>>>> Attendant. 2. Customer is presented with a menu and makes selection.
>>>>> His call is put on hold and an entry is made in my system's database
>>>>> for incoming queries. 3. These queries are shown to agents handling
>>>>> calls.
>>>>> 4. And agent clicks on a query, he is shown an extension where call is
>>>>> parked. He dials that and is connected to the customer.
>>>>>
>>>>> 5. He talks to the customer and resolve his queries.
>>>>>
>>>>> Please guide me on how can do it. The application will be written in
>>>>> Java.
>>>>>
>>>>> I am an experienced programmers in Java/J2EE; but doesn't have much
>>>>> knowledge of VoIP/FreeSWITCH. I've configured FreeSWITCH on Ubuntu
>>>>> 12.04; and have installed X-Lite Softphones on two Windows machines.
>>>>> I configured these phones to work with FreeSWITCH and they are
>>>>> working fine.
>>>>>
>>>>>
>>>>>
>>>>> I think that I'd have to use Valet Parking to park customer's call on
>>>>> an
>>>>> extension, then pass that extension to agent who will dial it and will
>>>>> be connected to the customer.
>>>>>
>>>>>
>>>>>
>>>>> Please tell me whether this approach will work? And how should I go
>>>>> about it.
>>>>>
>>>>>
>>>>>
>>>>> All help would be much appreciated.
>>>>>
>>>>>
>>>>>
>>>>> Regards
>>>>>
>>>>>
>>>>>
>>>>> Nitin
>>>> ________________________________________________________________________
>>>> _
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
>


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

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Tel?fono: +58 212 9081842
Cel: +58 412 2221515
dgarcia at anew.com.ve
http://www.anew.com.ve
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From brett at launch3.net  Mon Aug 20 20:31:09 2012
From: brett at launch3.net (Brett Wilson)
Date: Mon, 20 Aug 2012 12:31:09 -0400
Subject: [Freeswitch-users] Mod_callcenter - cannot blind transfer 1 legged
	calls
Message-ID: <001c01cd7ef1$34ca8c80$9e5fa580$@launch3.net>

Hey guys, 

I'm trying to implement mod callcenter, I've got everything setup (from
within fusionpbx) and I'm getting 'cannot blind transfer 1 legged calls'
when trying to send a call to the acd.

 

Log:

http://pastebin.com/y2U1g2f2

 

Also, an unrelated question - what is the normal way to set up a fs box
behind a firewall and on a local lan with regards to hostname and such? I
have the box setup on the dhcp server as always getting assigned ip
192.168.2.7 via mac address. I prefer this to setting static ip in the host
os. The box is sip registering to my voice provider. On the dhcp server, I
am also running a dns server that resolves some local lan hostnames and
forwards anything else to google or opendns. Also computer hostnames are
automatically added to the dns database, so if the hostname for the fs box
is 'hades', 'hades.l3office.net' will resolve to the fs box, but only from
within the lan. L3office.net does not actually exist and is not routable
from the wan. Where I am unclear is what to set the domain to in the fs
config, hades.l3office.net or 192.168.2.7? And in the phone's sip server? 

 

TIA for the assistance.

 

 

******************************************* 
Brett Wilson
IT Department
Launch 3 Ventures, LLC 
134 Myer Street 
Hackensack, NJ 07601 
Phone: 877.878.9134
Fax: 646.536.3866 
Email:  <mailto:Brett.Wilson at launch3.net> Brett.Wilson at launch3.net
AOL IM:  <mailto:Brett.Wilson at launch3.net> Brett.Wilson at launch3.net
 <http://www.launch3.net/> www.Launch3.net 
 <http://www.launch3telecom.com/> www.Launch3telecom.com 
******************************************* 
 <http://launch3telecom.blogspot.com/> Description: Description:
Description: Blogger-logo
<http://www.facebook.com/pages/Launch-3-Telecom/165610820178720>
Description: Description: Description: FaceBook-Logo
<https://twitter.com/launch3telecom> Description: Description: Description:
Twitter-Logo <https://plus.google.com/102713212642017290959> Description:
Description: Description: GPlus-Logo

 

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From g.d.monnezza at tiscali.it  Mon Aug 20 20:32:14 2012
From: g.d.monnezza at tiscali.it (g)
Date: Mon, 20 Aug 2012 18:32:14 +0200
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <50323D6A.3020708@anew.com.ve>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<503226CE.7040307@hw.ac.uk> <50323D6A.3020708@anew.com.ve>
Message-ID: <2526639.vT3cT5qGKG@virtex>

Thanks Dario. You gave the answer I would like to do.
Liu, You can avoid agents dialing extensions to login/logout/pause, but at 
least they have to click on some web app to let the system knows they are 
available.
Another option is to let the system considerthem always available even when 
the softphones are'nt registered. If you set a correct distribution criteria, 
the unregistered softphones are skipped in a round robin distribution, as the 
agents were not logged in. But in this case, forget about statistics on drops, 
presence, waiting time on queue etc.
g

On Monday 20 August 2012 09:06:42 Saugort Dario Garcia Tovar wrote:
> mmm,
> 
> You think that the agent is dialing an extension. It is not what is
> happening. The agent is dialing a number that invoke a mod_callcenter
> function: login/logout/not ready. As FS is like a pbx (and more than a
> pbx), it seems like an extension.
> 
> Just, registrer to FS is not enough. What happen when the agent need to
> go the restroom, take coffee, couching, or take a break? Then, you will
> need to reproduce in the agent gui the funcion of login, logout and
> break... but again, what happen if your agent gui fail? your callcenter
> service will stop because your gui fail? Also, if you just use register
> signaling, you could face a wide sort of operational issues like, what
> happen if the agent left the position and left the their seats: then
> calls will be routed to empty seats!
> 
> If you want to get the agent login/logout without the agent dial
> nothing... if I am not wrong, mod_callcenter offer functions to log and
> logout agent from fs console, so I think you can invoke them using
> execute() from lua or your script.
> 
> On 8/20/2012 7:30 AM, x.liu wrote:
> > Hi g,
> > 
> > Thanks for your response!
> > 
> > My situation is that I don't want the agent to dial the extension to be
> > able to accept calls.
> > The agents (e.g. a FS softphone client, or a Java application) just need
> > to register to FS as SIP clients.
> > 
> > And I've not figured out what commands I should send if I use extension
> > "^agent-login$" and FS socket.
> > 
> > So I guess I may be better not to use mod_callcenter for routing calls
> > to different extensions.
> > just to use FS ESL and my own logics to bridge the calls.
> > 
> > I was curious about the reply "You don't need agent to dial anything to
> > get the call from queue... "
> > and I thought my question is relevant to the thread, so I asked
> > questions in this thread.
> > Apologies for that. I could have opened a different thread for the
> > questions :-)
> > 
> > Cheers,
> > Xing
> > 
> > On 08/20/2012 11:53 AM, g wrote:
> >> You can log an agent into a queue by a web application in which a
> >> button click handles commands via the FS socket, or just let the
> >> agent dial an extension (i.e. 999+itssoftphonenumber) where dialplan
> >> sets that agent in the correct state.
> >> ie (replace agent_login regex into something like 999201 if your agent
> >> is 201) <extension name="agent_login">
> >> 
> >>     <condition field="destination_number"
> >>     expression="^agent-login$">
> >>     
> >>       <action application="set"
> >>       data="res=${callcenter_config(agent set status>> 
> >> ${caller_id_number}@${domain_name} 'Available')}" />
> >> 
> >>       <action application="answer" data=""/>
> >>       <action application="sleep" data="500"/>
> >>       <action application="playback"
> >>       data="ivr/ivr-you_are_now_logged_in.wav"/> <action
> >>       application="hangup" data=""/>
> >>     
> >>     </condition>
> >> 
> >> </extension>
> >> 
> >> <extension name="agent_logoff">
> >> 
> >>     <condition field="destination_number"
> >>     expression="^agent-logoff$">
> >>     
> >>       <action application="set"
> >>       data="res=${callcenter_config(agent set status>> 
> >> ${caller_id_number}@${domain_name} 'Logged Out')}" />
> >> 
> >>       <action application="answer" data=""/>
> >>       <action application="sleep" data="500"/>
> >>       <action application="playback"
> >>       data="ivr/ivr-you_are_now_logged_out.wav"/> <action
> >>       application="hangup" data=""/>
> >>     
> >>     </condition>
> >> 
> >> </extension>
> >> 
> >> The important thing is tha the agent is, somehow, putted in
> >> "available"
> >> status. You can also do it via fs_cli, but now I can't remeber the
> >> exact
> >> syntax for that. Help from cli should clarify.
> >> 
> >> At this point, the agent should result in "waiting" state
> >> Of course, you should before declared that agent as member of a tier
> >> or, at least, of a queue.
> >> 
> >> Hope it helps
> >> g
> >> 
> >> On Friday 17 August 2012 14:48:04 x.liu wrote:
> >>> I am also trying the mod_callcenter. What I want is to route a call
> >>> to
> >>> one agent ( a registered FS client, e.g. a SIP softphone)
> >>> of a list of agents by a Random or RoundRobin policy.
> >>> 
> >>> When I dial the extension which calls the callcenter app, I only
> >>> have
> >>> hold-on music. It does not route a call to a agent.
> >>> You said " You don't need agent to dial anything to get the call
> >>> from
> >>> queue. ",
> >>> I am wondering how I could let an agent log into the queue?
> >>> 
> >>> Thanks!
> >>> 
> >>> Xing
> >>> 
> >>> On 08/17/2012 01:56 PM, g wrote:
> >>>> Me too, I'm often dealing with callcenters (inbound+outbound). I
> >>>> found
> >>>> in
> >>>> Freeswitch the easyest solution to manage queue and agents. I'm
> >>>> not a
> >>>> programmer, but I have experience with Asterisk, Freeswitch and
> >>>> the
> >>>> whole
> >>>> workflow of complex callcenters.
> >>>> My suggestion too is to use mod_callcenter. Easy, fast, powerful.
> >>>> You
> >>>> don't need agent to dial anything to get the call from queue.
> >>>> In my opinion, you can use that module or, for more complex things
> >>>> (ie. outbound predictive) you can use the socket manager to
> >>>> perform  call placement, call transfer, conference with
> >>>> thirdy-part recording  IVR etc.
> >>>> I feel the best solution, also if it will require some effort for
> >>>> a Java programmer, is to do all that in php, which is light and
> >>>> fast also for older PC, able to run also with thin-client low
> >>>> memory low-cpu machines. If you can plan to turn to php and you
> >>>> don't have too short time, I can consider to help for the
> >>>> Freeswitch part.
> >>>> giuliano
> >>>> 
> >>>> On Thursday 16 August 2012 17:10:33 Nitin Tomer wrote:
> >>>>> Hi,
> >>>>> 
> >>>>> 
> >>>>> 
> >>>>> I am working on a Contact Center solution. It will support mail,
> >>>>> chat
> >>>>> and call queries.
> >>>>> 
> >>>>> The requirements are:
> >>>>> 
> >>>>> 1. An end-customer calls, the call is handled by FreeSWITCH Auto
> >>>>> Attendant. 2. Customer is presented with a menu and makes
> >>>>> selection.
> >>>>> His call is put on hold and an entry is made in my system's
> >>>>> database
> >>>>> for incoming queries. 3. These queries are shown to agents
> >>>>> handling
> >>>>> calls.
> >>>>> 4. And agent clicks on a query, he is shown an extension where
> >>>>> call is parked. He dials that and is connected to the customer.
> >>>>> 
> >>>>> 5. He talks to the customer and resolve his queries.
> >>>>> 
> >>>>> Please guide me on how can do it. The application will be
> >>>>> written in
> >>>>> Java.
> >>>>> 
> >>>>> I am an experienced programmers in Java/J2EE; but doesn't have
> >>>>> much
> >>>>> knowledge of VoIP/FreeSWITCH. I've configured FreeSWITCH on
> >>>>> Ubuntu
> >>>>> 12.04; and have installed X-Lite Softphones on two Windows
> >>>>> machines.
> >>>>> I configured these phones to work with FreeSWITCH and they are
> >>>>> working fine.
> >>>>> 
> >>>>> 
> >>>>> 
> >>>>> I think that I'd have to use Valet Parking to park customer's
> >>>>> call on
> >>>>> an
> >>>>> extension, then pass that extension to agent who will dial it
> >>>>> and will be connected to the customer.
> >>>>> 
> >>>>> 
> >>>>> 
> >>>>> Please tell me whether this approach will work? And how should I
> >>>>> go
> >>>>> about it.
> >>>>> 
> >>>>> 
> >>>>> 
> >>>>> All help would be much appreciated.
> >>>>> 
> >>>>> 
> >>>>> 
> >>>>> Regards
> >>>>> 
> >>>>> 
> >>>>> 
> >>>>> Nitin
> >>>> 
> >>>> __________________________________________________________________
> >>>> ______ _
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>> 
> >>>> 
> >>>> 
> >>>> 
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.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
> >> 
> >> ______________________________________________________________________
> >> ___ Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >> 
> >> 
> >> 
> >> 
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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-use
> >> rs
> >> http://www.freeswitch.org


From marketing at cluecon.com  Mon Aug 20 21:56:21 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 20 Aug 2012 10:56:21 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxXKCQnH+cDshF95FruwSFTxTyc-rXq8Ww81mXXbJe-2tA@mail.gmail.com>

FreeSWITCH Weekly News and Notes is back after a brief hiatus.

In case you hadn't heard: FreeSWITCH 1.2 is out!<http://freeswitch.org/node/410>
 In fact, Anthony and Ken are working on a 1.2.2 release. Stay tuned for an
announcement. On last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_15>
 we discussed some of the git commands you may need to run in order to get
yourself moved up to the 1.2stable
branch<http://fisheye.freeswitch.org/changelog/~br=v1.2.stable/freeswitch.git>.
Join us this Wednesday<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22>
 and we'll do a quick follow up for those who may still have questions. We
hope to have some other announcements as well.

In post-ClueCon news we'd like to let everyone know that
Vestec<http://vestec.com/>
 is finalizing the arrangements for the great ASR (automated speech
recognition) app-building contest. This is a great opportunity to get some
cash and free speech recognition licenses in return for investing some time
and effort into learning the Vestec system and building an application to
show off to the world. It's also a great way to help promote FreeSWITCH
among larger enterprises who may not realize that professional-grade ASR is
available. We will discuss this further on Wednesday's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22>
.

Lastly we'd like to let everyone know that the ClueCon videos will be made
available in the coming weeks and months. Please give us some time to do a
little editing before we release them all. It will be worth the wait!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From brett at launch3.net  Mon Aug 20 23:33:49 2012
From: brett at launch3.net (Brett Wilson)
Date: Mon, 20 Aug 2012 15:33:49 -0400
Subject: [Freeswitch-users] mod_portaudio cannot find input
	or	output	device
In-Reply-To: <020301cd7cb8$d5985790$80c906b0$@launch3.net>
References: <021301cd7c72$6adff2f0$409fd8d0$@launch3.net>	<502E3CE7.4070902@puzzled.xs4all.nl>	<01e001cd7c9c$3c454f10$b4cfed30$@launch3.net>	<CALn233r1Y0QfHzkT7fU_N8neSztcMLhx38UR2ZnjNy_Xydiejg@mail.gmail.com>
	<020301cd7cb8$d5985790$80c906b0$@launch3.net>
Message-ID: <00ea01cd7f0a$b9474a70$2bd5df50$@launch3.net>

Bump. Anybody got any more info for me?

*******************************************?
Brett Wilson
IT Department
Launch 3 Ventures, LLC?
134 Myer Street?
Hackensack, NJ 07601?
Phone:?877.878.9134
Fax:?646.536.3866?
Email:?Brett.Wilson at launch3.net
AOL IM:?Brett.Wilson at launch3.net
www.Launch3.net?
www.Launch3telecom.com?
*******************************************?


-----Original Message-----
From: Brett Wilson [mailto:brett at launch3.net] 
Sent: Friday, August 17, 2012 4:43 PM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] mod_portaudio cannot find input or output
device

root     16791  1.6  0.9  37076 14456 ?        S<l  16:33   0:08
/usr/local/freeswitch/bin/freeswitch -rp -nc

*******************************************
Brett Wilson
IT Department
Launch 3 Ventures, LLC
134 Myer Street
Hackensack, NJ 07601
Phone:?877.878.9134
Fax:?646.536.3866
Email:?Brett.Wilson at launch3.net
AOL IM:?Brett.Wilson at launch3.net
www.Launch3.net
www.Launch3telecom.com
*******************************************?



-----Original Message-----
From: Mitch Capper [mailto:mitch.capper at gmail.com]
Sent: Friday, August 17, 2012 1:39 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_portaudio cannot find input or output
device

in/outdev dont matter as portaudio is saying it cant find any device.
Paste the results of:
ps auxf | grep freeswitch

back.

~Mitch

On Fri, Aug 17, 2012 at 10:17 AM, Brett Wilson <brett at launch3.net> wrote:
> I disabled loading of any modules besides the bare miniumum, still no
dice.
> I set the in/out device as #0 as shown with aplay -l. Still no good.
>
> *******************************************
> Brett Wilson
> IT Department
> Launch 3 Ventures, LLC
> 134 Myer Street
> Hackensack, NJ 07601
> Phone: 877.878.9134
> Fax: 646.536.3866
> Email: Brett.Wilson at launch3.net
> AOL IM: Brett.Wilson at launch3.net
> www.Launch3.net
> www.Launch3telecom.com
> *******************************************
>
>
>
> -----Original Message-----
> From: Patrick Lists [mailto:freeswitch-list at puzzled.xs4all.nl]
> Sent: Friday, August 17, 2012 8:45 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_portaudio cannot find input or 
> output device
>
> On 17-08-12 14:18, Brett Wilson wrote:
> [snip]
>> *freeswitch at internal> load mod_portaudio*
>>
>> +OK Reloading XML
>>
>> -ERR [module load file routine returned an error]
>
> That ERR does not look good. Iirc I have seen that with mod_java when 
> starting FreeSWITCH. Subsequent manual loading of the mod_java module 
> went fine. But in your case you are already manually loading the module.
>
>> *2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1767 Cannot find an 
>> input device*
>>
>> *2012-08-17 08:15:23.591770 [ERR] mod_portaudio.c:1777 Cannot find an 
>> output device*
>
> Did you set the input/output devices? Lot's of info on the wiki:
>
> http://wiki.freeswitch.org/wiki/FreeSwitch_Enpoint_Portaudio#List_the_
> availa
> ble_devices
>
> Regards,
> Patrick
>
>
>
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-use
> rs
> http://www.freeswitch.org








From freeswitch-list at puzzled.xs4all.nl  Tue Aug 21 00:00:43 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Mon, 20 Aug 2012 22:00:43 +0200
Subject: [Freeswitch-users] mod_portaudio cannot find input or	output
 device
In-Reply-To: <00ea01cd7f0a$b9474a70$2bd5df50$@launch3.net>
References: <021301cd7c72$6adff2f0$409fd8d0$@launch3.net>	<502E3CE7.4070902@puzzled.xs4all.nl>	<01e001cd7c9c$3c454f10$b4cfed30$@launch3.net>	<CALn233r1Y0QfHzkT7fU_N8neSztcMLhx38UR2ZnjNy_Xydiejg@mail.gmail.com>
	<020301cd7cb8$d5985790$80c906b0$@launch3.net>
	<00ea01cd7f0a$b9474a70$2bd5df50$@launch3.net>
Message-ID: <5032976B.7050208@puzzled.xs4all.nl>

On 20-08-12 21:33, Brett Wilson wrote:
> Bump. Anybody got any more info for me?

Not sure if mentioned before but have you tried disabling SELinux?

Regards,
Patrick




From onyeagbaikenna04 at gmail.com  Mon Aug 20 23:57:30 2012
From: onyeagbaikenna04 at gmail.com (Ikenna Onyeagba)
Date: Mon, 20 Aug 2012 20:57:30 +0100
Subject: [Freeswitch-users] make && make install error
Message-ID: <CAHibNNfMLLg4CtG5bJKq8fBwq_U4VeQM_npOYRpedHOWQ24k=Q@mail.gmail.com>

i get this error when i run the make && make install error.
Making all in doc
make[3]: Entering directory
`/home/free/usr/src/freeswitch-1.0.6/libs/libedit/doc'
if test "mdoc" = "mdoc"; then\
   cp ./editline.3.roff editline.3;\
        else\
   gawk -f ./mdoc2man.awk ./editline.3.roff > editline.3 || rm -f editline.3;\
fi;
if test "mdoc" = "mdoc"; then\
   cp ./editrc.5.roff editrc.5;\
        else\
   gawk -f ./mdoc2man.awk ./editrc.5.roff > editrc.5 || rm -f editrc.5;\
fi;
make[3]: Leaving directory
`/home/free/usr/src/freeswitch-1.0.6/libs/libedit/doc'
make[3]: Entering directory `/home/free/usr/src/freeswitch-1.0.6/libs/libedit'
make[3]: Leaving directory `/home/free/usr/src/freeswitch-1.0.6/libs/libedit'
make[2]: Leaving directory `/home/free/usr/src/freeswitch-1.0.6/libs/libedit'
make[1]: Leaving directory `/home/free/usr/src/freeswitch-1.0.6/libs/libedit'
gcc -E /home/free/usr/src/freeswitch-1.0.6/src/include/switch_cpp.h
-DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x
-DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x
2>/dev/null | grep -v "^#" > src/include/switch_swigable_cpp.h
make: *** [src/include/switch_swigable_cpp.h] Error 1
[root at Ikenna-TOSH freeswitch-1.0.6]#
pls i need help on how to resolve this problem urgently. thanks in advance
Ikenna


From lloyd.aloysius at gmail.com  Tue Aug 21 00:35:18 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Mon, 20 Aug 2012 16:35:18 -0400
Subject: [Freeswitch-users] make && make install error
In-Reply-To: <CAHibNNfMLLg4CtG5bJKq8fBwq_U4VeQM_npOYRpedHOWQ24k=Q@mail.gmail.com>
References: <CAHibNNfMLLg4CtG5bJKq8fBwq_U4VeQM_npOYRpedHOWQ24k=Q@mail.gmail.com>
Message-ID: <CAGvuWYZZMyOP2N8N5+hDWbdMZUNrHdUjvTSAZJ3NXMV7PAVzzg@mail.gmail.com>

when I look into the error ... the path showing the 1.0.6

*/home/free/usr/src/freeswitch-1.0.6/ *

Are you using the old version?


On Mon, Aug 20, 2012 at 3:57 PM, Ikenna Onyeagba <onyeagbaikenna04 at gmail.com
> wrote:

> i get this error when i run the make && make install error.
> Making all in doc
> make[3]: Entering directory
> `/home/free/usr/src/freeswitch-1.0.6/libs/libedit/doc'
> if test "mdoc" = "mdoc"; then\
>    cp ./editline.3.roff editline.3;\
>         else\
>    gawk -f ./mdoc2man.awk ./editline.3.roff > editline.3 || rm -f
> editline.3;\
> fi;
> if test "mdoc" = "mdoc"; then\
>    cp ./editrc.5.roff editrc.5;\
>         else\
>    gawk -f ./mdoc2man.awk ./editrc.5.roff > editrc.5 || rm -f editrc.5;\
> fi;
> make[3]: Leaving directory
> `/home/free/usr/src/freeswitch-1.0.6/libs/libedit/doc'
> make[3]: Entering directory
> `/home/free/usr/src/freeswitch-1.0.6/libs/libedit'
> make[3]: Leaving directory
> `/home/free/usr/src/freeswitch-1.0.6/libs/libedit'
> make[2]: Leaving directory
> `/home/free/usr/src/freeswitch-1.0.6/libs/libedit'
> make[1]: Leaving directory
> `/home/free/usr/src/freeswitch-1.0.6/libs/libedit'
> gcc -E /home/free/usr/src/freeswitch-1.0.6/src/include/switch_cpp.h
> -DSWITCH_DECLARE_CLASS= -DSWITCH_DECLARE\(x\)=x
> -DSWITCH_DECLARE_CONSTRUCTOR= -DSWITCH_DECLARE_NONSTD\(x\)=x
> 2>/dev/null | grep -v "^#" > src/include/switch_swigable_cpp.h
> make: *** [src/include/switch_swigable_cpp.h] Error 1
> [root at Ikenna-TOSH freeswitch-1.0.6]#
> pls i need help on how to resolve this problem urgently. thanks in advance
> Ikenna
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 brett at launch3.net  Tue Aug 21 00:51:20 2012
From: brett at launch3.net (Brett Wilson)
Date: Mon, 20 Aug 2012 16:51:20 -0400
Subject: [Freeswitch-users] mod_portaudio cannot find input
	or	output	device
In-Reply-To: <5032976B.7050208@puzzled.xs4all.nl>
References: <021301cd7c72$6adff2f0$409fd8d0$@launch3.net>	<502E3CE7.4070902@puzzled.xs4all.nl>	<01e001cd7c9c$3c454f10$b4cfed30$@launch3.net>	<CALn233r1Y0QfHzkT7fU_N8neSztcMLhx38UR2ZnjNy_Xydiejg@mail.gmail.com>	<020301cd7cb8$d5985790$80c906b0$@launch3.net>	<00ea01cd7f0a$b9474a70$2bd5df50$@launch3.net>
	<5032976B.7050208@puzzled.xs4all.nl>
Message-ID: <011501cd7f15$8d9315c0$a8b94140$@launch3.net>

It is not installed.

*******************************************?
Brett Wilson
IT Department
Launch 3 Ventures, LLC?
134 Myer Street?
Hackensack, NJ 07601?
Phone:?877.878.9134
Fax:?646.536.3866?
Email:?Brett.Wilson at launch3.net
AOL IM:?Brett.Wilson at launch3.net
www.Launch3.net?
www.Launch3telecom.com?
*******************************************?



-----Original Message-----
From: Patrick Lists [mailto:freeswitch-list at puzzled.xs4all.nl] 
Sent: Monday, August 20, 2012 4:01 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_portaudio cannot find input or output
device

On 20-08-12 21:33, Brett Wilson wrote:
> Bump. Anybody got any more info for me?

Not sure if mentioned before but have you tried disabling SELinux?

Regards,
Patrick







From brett at launch3.net  Tue Aug 21 00:55:53 2012
From: brett at launch3.net (Brett Wilson)
Date: Mon, 20 Aug 2012 16:55:53 -0400
Subject: [Freeswitch-users] CIDlookup
Message-ID: <011701cd7f16$302595b0$9070c110$@launch3.net>

I am trying to get CID lookup to display on the phone's display. I am using
this code in the beginning of the public.xml dialplan:

 

<extension name="cid_number_cleanup" continue="true">

      <condition field="caller_id_number" expression="^(?:\+)(\d+)$">

        <action application="set" data="effective_caller_id_number=$1"
inline="true"/>

      </condition>

    </extension>

 

    <extension name="cid_name_cleanup" continue="true">

      <condition field="caller_id_name" expression="^(?:\+)(\d+)$">

        <action application="set" data="effective_caller_id_name=$1"
inline="true"/>

      </condition>

    </extension>

 

    <extension name="cid_lookup-country_code_1" continue="true">

      <condition field="${module_exists(mod_cidlookup)}" expression="true"/>

      <condition field="caller_id_name" expression="^(?:\+)(\d+)$|^$"/>

      <condition field="caller_id_number"
expression="^(?:\+1|1)?([2-9]\d\d[2-9]\d{6})$">

        <action application="cidlookup" data="$1"/>

      </condition>

    </extension>

 

The lookup works - I can see the result in the cdr records. But it does not
display on the phone's display. When we get a call in with actual cnam info,
that does display on the phone. The difference I think is when we get a cnam
call, sip_from_display is filled with the name. When we do the cid lookup,
it is not filled. I tried to set sip_from_display myself through channel
variables in the bridge string as well as export but it is not working. Any
ideas on how I can do this? Being able to control what is displayed on the
screen would be ideal so I can let users know when they are getting a call
direct to their extension or an ACD call.

 

TIA.

 

******************************************* 
Brett Wilson
IT Department
Launch 3 Ventures, LLC 
134 Myer Street 
Hackensack, NJ 07601 
Phone: 877.878.9134
Fax: 646.536.3866 
Email:  <mailto:Brett.Wilson at launch3.net> Brett.Wilson at launch3.net
AOL IM:  <mailto:Brett.Wilson at launch3.net> Brett.Wilson at launch3.net
 <http://www.launch3.net/> www.Launch3.net 
 <http://www.launch3telecom.com/> www.Launch3telecom.com 
******************************************* 
 <http://launch3telecom.blogspot.com/> Description: Description:
Description: Blogger-logo
<http://www.facebook.com/pages/Launch-3-Telecom/165610820178720>
Description: Description: Description: FaceBook-Logo
<https://twitter.com/launch3telecom> Description: Description: Description:
Twitter-Logo <https://plus.google.com/102713212642017290959> Description:
Description: Description: GPlus-Logo

 

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From onyeagbaikenna04 at gmail.com  Tue Aug 21 00:46:57 2012
From: onyeagbaikenna04 at gmail.com (onyeagbaikenna04 at gmail.com)
Date: Mon, 20 Aug 2012 20:46:57 +0000
Subject: [Freeswitch-users] make && make install error
In-Reply-To: <CAGvuWYZZMyOP2N8N5+hDWbdMZUNrHdUjvTSAZJ3NXMV7PAVzzg@mail.gmail.com>
References: <CAHibNNfMLLg4CtG5bJKq8fBwq_U4VeQM_npOYRpedHOWQ24k=Q@mail.gmail.com>
	<CAGvuWYZZMyOP2N8N5+hDWbdMZUNrHdUjvTSAZJ3NXMV7PAVzzg@mail.gmail.com>
Message-ID: <207711913-1345496318-cardhu_decombobulator_blackberry.rim.net-289830855-@b13.c16.bise7.blackberry>

I don't knw if FS has a later version than 1.0.6. What can I do to counter the error I get wen I run the make && make install command

Sent from my BlackBerry smartphone from Virgin Media

-----Original Message-----
From: Lloyd Aloysius <lloyd.aloysius at gmail.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Mon, 20 Aug 2012 16:35:18 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] make && make install error

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 lloyd.aloysius at sunteltech.ca  Tue Aug 21 01:15:00 2012
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Mon, 20 Aug 2012 17:15:00 -0400
Subject: [Freeswitch-users] make && make install error
In-Reply-To: <207711913-1345496318-cardhu_decombobulator_blackberry.rim.net-289830855-@b13.c16.bise7.blackberry>
References: <CAHibNNfMLLg4CtG5bJKq8fBwq_U4VeQM_npOYRpedHOWQ24k=Q@mail.gmail.com>
	<CAGvuWYZZMyOP2N8N5+hDWbdMZUNrHdUjvTSAZJ3NXMV7PAVzzg@mail.gmail.com>
	<207711913-1345496318-cardhu_decombobulator_blackberry.rim.net-289830855-@b13.c16.bise7.blackberry>
Message-ID: <CAGvuWYZM2aFg7tRwens1icFepVCw91Edo7sTt2S4tSjEXe508g@mail.gmail.com>

see below . You should use 1.2.stable from git.

http://freeswitch.org/node/410

*
*
*  <http://www.sunteltech.ca/blog/>*



On Mon, Aug 20, 2012 at 4:46 PM, <onyeagbaikenna04 at gmail.com> wrote:

> I don't knw if FS has a later version than 1.0.6. What can I do to counter
> the error I get wen I run the make && make install command
>
> Sent from my BlackBerry smartphone from Virgin Media
>
> -----Original Message-----
> From: Lloyd Aloysius <lloyd.aloysius at gmail.com>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Mon, 20 Aug 2012 16:35:18
> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] make && make install error
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Aug 21 01:45:31 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Aug 2012 14:45:31 -0700
Subject: [Freeswitch-users] FreeSWITCH Project Looking For Volunteers: Web
	Site Dev/Maint
Message-ID: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>

Hello all,

I need to refresh my list of people who have Web dev skills and are willing
to assist with things like Web site design, programming, and maintenance.
If you have any of the following skills, or know someone who does and who
is willing to donate a few hours, please email me offlist:

HTML5
CSS2/3
Javascript/JQuery
Drupal/PHP
Wordpress/PHP
Django/Python
Web design/graphics
Apache administration in LAMP environment

Please note that not every single skill is currently in demand; we are
trying to get a feel for who knows what and hopefully anticipate future
needs.

Thanks!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From mitch.capper at gmail.com  Tue Aug 21 02:28:21 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 20 Aug 2012 15:28:21 -0700
Subject: [Freeswitch-users] mod_portaudio cannot find input or output
	device
In-Reply-To: <011501cd7f15$8d9315c0$a8b94140$@launch3.net>
References: <021301cd7c72$6adff2f0$409fd8d0$@launch3.net>
	<502E3CE7.4070902@puzzled.xs4all.nl>
	<01e001cd7c9c$3c454f10$b4cfed30$@launch3.net>
	<CALn233r1Y0QfHzkT7fU_N8neSztcMLhx38UR2ZnjNy_Xydiejg@mail.gmail.com>
	<020301cd7cb8$d5985790$80c906b0$@launch3.net>
	<00ea01cd7f0a$b9474a70$2bd5df50$@launch3.net>
	<5032976B.7050208@puzzled.xs4all.nl>
	<011501cd7f15$8d9315c0$a8b94140$@launch3.net>
Message-ID: <CALn233qreDcwqDrB20EKDZK1nHCo2SeiO4U68evhK_7WZu=hsA@mail.gmail.com>

I am not sure why portaudio external can see devices just fine but the
FS portaudio build can't unfortunately you would need to debug in the
code most likely to figure out further, I have no further tips as to
why it can't find any devices.

~Mitch

On Mon, Aug 20, 2012 at 1:51 PM, Brett Wilson <brett at launch3.net> wrote:
> It is not installed.
>
> *******************************************
> Brett Wilson
> IT Department
> Launch 3 Ventures, LLC
> 134 Myer Street
> Hackensack, NJ 07601
> Phone: 877.878.9134
> Fax: 646.536.3866
> Email: Brett.Wilson at launch3.net
> AOL IM: Brett.Wilson at launch3.net
> www.Launch3.net
> www.Launch3telecom.com
> *******************************************
>
>
>
> -----Original Message-----
> From: Patrick Lists [mailto:freeswitch-list at puzzled.xs4all.nl]
> Sent: Monday, August 20, 2012 4:01 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_portaudio cannot find input or output
> device
>
> On 20-08-12 21:33, Brett Wilson wrote:
>> Bump. Anybody got any more info for me?
>
> Not sure if mentioned before but have you tried disabling SELinux?
>
> Regards,
> Patrick
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lists at kavun.ch  Tue Aug 21 02:56:52 2012
From: lists at kavun.ch (Emrah)
Date: Mon, 20 Aug 2012 18:56:52 -0400
Subject: [Freeswitch-users] Regex issue
In-Reply-To: <CADEQS5y=-Ur5WYeBR5TW=BC2-K447KtpgfjfE__OU9NECQc9Fg@mail.gmail.com>
References: <C92661B1-1476-4FBE-9181-34DEAD7AADC7@kavun.ch>
	<CADEQS5y=-Ur5WYeBR5TW=BC2-K447KtpgfjfE__OU9NECQc9Fg@mail.gmail.com>
Message-ID: <485910E8-ADEC-4D97-A6BF-5252E28C5FAE@kavun.ch>

Hi Brian,

Thanks a million for your response. I was asuming that the variable would be automagically expanded. 

My emails take ages to make it to the ML and I've clocked in quite some FS miles since I posted this.

Best to you,
Emrah
On Aug 19, 2012, at 1:14 PM, Brian Wiese <brian.wiese.freeswitch at gmail.com> wrote:

> Emrah:
> 
> Try <regex field="${allow_311}" expression="^true$" />
> 
> If that doesn't work please pastebin a log of a call.
> 
> ~Brian
> 
> On Sun, Aug 19, 2012 at 12:01 PM, Emrah <lists at kavun.ch> wrote:
>> Hi all,
>> 
>> I am trying to e164ize the caller id number received on my inbound dids and
>> thought I would experiment with multiple conditions first. My test is to
>> only allow user 20 and 21 to call 311 in NYC. Note that I already have 311
>> working by using another evaluation process.
>> 
>> I know I can optimize the caller id conditions by piping each possibility,
>> but it's not what's not working. I'd like to keep it that way for
>> experimentation purposes.
>> 
>> The line that I suspect causing the issue is highlighted in my pastebin
>> post, available here:
>> http://pastebin.freeswitch.org/19735
>> 
>> Your help would be greatly appreciated.
>> 
>> thanks,
>> 
>> --
>> Emrah
>> 
>> ?In theory, theory and practice are the same. In practice, they are not.?
>> Albert Einstein
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 info at pripojtese.net  Tue Aug 21 03:11:11 2012
From: info at pripojtese.net (Jakub Tencl)
Date: Tue, 21 Aug 2012 00:11:11 +0100
Subject: [Freeswitch-users] Help with settings
Message-ID: <5032C40F.9050302@pripojtese.net>

Hello,

is here anyone who is willing to help me with a few things on my 
freeswitch server remotely?

Thanks






From fernandojdk at gmail.com  Tue Aug 21 03:44:52 2012
From: fernandojdk at gmail.com (Fernando - NextBilling IP Solutions)
Date: Mon, 20 Aug 2012 18:44:52 -0500
Subject: [Freeswitch-users] FreeSWITCH Project Looking For Volunteers:
	Web Site Dev/Maint
In-Reply-To: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
Message-ID: <16354AFE-5296-452A-81A2-E6181FA74AB9@gmail.com>

Hello.

I'm ready for:

HTML5
CSS2/3
Javascript/jQuery
Advanced PHP
Apache LAMP enviroment Admin 

Atenciosamente,





Importante:
Esta mensagem, incluindo todo seu conte?do, cont?m informa??es confidenciais, legalmente protegidas e destinadas a indiv?duo e prop?sito espec?ficos.
Caso a tenha recebido por engano, lembramos do car?ter sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o mais breve poss?vel.
As informa??es contidas nesta mensagem e em seu conte?do s?o de responsabilidade de seu autor, n?o representando necessariamente id?ias, opini?es, pensamentos ou qualquer forma de posicionamento por parte da NextBilling IP Solutions.
P "Antes de imprimir pense em seu compromisso com o Meio Ambiente."

Atenciosamente,





Importante:
Esta mensagem, incluindo todo seu conte?do, cont?m informa??es confidenciais, legalmente protegidas e destinadas a indiv?duo e prop?sito espec?ficos.
Caso a tenha recebido por engano, lembramos do car?ter sigiloso e solicitamos a gentileza de desconsider?-la e comunicar-nos o mais breve poss?vel.
As informa??es contidas nesta mensagem e em seu conte?do s?o de responsabilidade de seu autor, n?o representando necessariamente id?ias, opini?es, pensamentos ou qualquer forma de posicionamento por parte da NextBilling IP Solutions.
P "Antes de imprimir pense em seu compromisso com o Meio Ambiente."

Em 08/20/2012, ?s 16:45, Michael Collins <msc at freeswitch.org> escreveu:

> Hello all,
> 
> I need to refresh my list of people who have Web dev skills and are willing to assist with things like Web site design, programming, and maintenance. If you have any of the following skills, or know someone who does and who is willing to donate a few hours, please email me offlist:
> 
> HTML5
> CSS2/3
> Javascript/JQuery
> Drupal/PHP
> Wordpress/PHP
> Django/Python
> Web design/graphics
> Apache administration in LAMP environment
> 
> Please note that not every single skill is currently in demand; we are trying to get a feel for who knows what and hopefully anticipate future needs.
> 
> Thanks!
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 brett at launch3.net  Tue Aug 21 03:56:47 2012
From: brett at launch3.net (Brett Wilson)
Date: Mon, 20 Aug 2012 19:56:47 -0400
Subject: [Freeswitch-users] mod_portaudio cannot find input or
	output	device
In-Reply-To: <CALn233qreDcwqDrB20EKDZK1nHCo2SeiO4U68evhK_7WZu=hsA@mail.gmail.com>
References: <021301cd7c72$6adff2f0$409fd8d0$@launch3.net>	<502E3CE7.4070902@puzzled.xs4all.nl>	<01e001cd7c9c$3c454f10$b4cfed30$@launch3.net>	<CALn233r1Y0QfHzkT7fU_N8neSztcMLhx38UR2ZnjNy_Xydiejg@mail.gmail.com>	<020301cd7cb8$d5985790$80c906b0$@launch3.net>	<00ea01cd7f0a$b9474a70$2bd5df50$@launch3.net>	<5032976B.7050208@puzzled.xs4all.nl>	<011501cd7f15$8d9315c0$a8b94140$@launch3.net>
	<CALn233qreDcwqDrB20EKDZK1nHCo2SeiO4U68evhK_7WZu=hsA@mail.gmail.com>
Message-ID: <06fb01cd7f2f$75b2b950$61182bf0$@launch3.net>

How do I go about debugging it if there is no crash? I looked at the wiki
but looks like gdb would only work if there is a crash.

*******************************************?
Brett Wilson
IT Department
Launch 3 Ventures, LLC?
134 Myer Street?
Hackensack, NJ 07601?
Phone:?877.878.9134
Fax:?646.536.3866?
Email:?Brett.Wilson at launch3.net
AOL IM:?Brett.Wilson at launch3.net
www.Launch3.net?
www.Launch3telecom.com?
*******************************************?


-----Original Message-----
From: Mitch Capper [mailto:mitch.capper at gmail.com] 
Sent: Monday, August 20, 2012 6:28 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_portaudio cannot find input or output
device

I am not sure why portaudio external can see devices just fine but the FS
portaudio build can't unfortunately you would need to debug in the code most
likely to figure out further, I have no further tips as to why it can't find
any devices.

~Mitch

On Mon, Aug 20, 2012 at 1:51 PM, Brett Wilson <brett at launch3.net> wrote:
> It is not installed.
>
> *******************************************
> Brett Wilson
> IT Department
> Launch 3 Ventures, LLC
> 134 Myer Street
> Hackensack, NJ 07601
> Phone: 877.878.9134
> Fax: 646.536.3866
> Email: Brett.Wilson at launch3.net
> AOL IM: Brett.Wilson at launch3.net
> www.Launch3.net
> www.Launch3telecom.com
> *******************************************
>
>
>
> -----Original Message-----
> From: Patrick Lists [mailto:freeswitch-list at puzzled.xs4all.nl]
> Sent: Monday, August 20, 2012 4:01 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_portaudio cannot find input or 
> output device
>
> On 20-08-12 21:33, Brett Wilson wrote:
>> Bump. Anybody got any more info for me?
>
> Not sure if mentioned before but have you tried disabling SELinux?
>
> Regards,
> Patrick
>
>
>
>
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-use
> rs
> http://www.freeswitch.org





From freeswitch at scottisheyes.com  Tue Aug 21 04:32:15 2012
From: freeswitch at scottisheyes.com (James)
Date: Mon, 20 Aug 2012 17:32:15 -0700
Subject: [Freeswitch-users] FreeSWITCH Project Looking For Volunteers:
 Web Site Dev/Maint
In-Reply-To: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
Message-ID: <CALfx8k1i79u5zuhxyZy-QtZba5p9+zkk6fFLkw2RFy+K3K_uqQ@mail.gmail.com>

I've got Django/Python, HTML, CSS, jQuery along with some server admin
knowledge.  I'll help if I can, time dependent (I'm a new dad to twins, and
yes, I like announcing it).

Cheers,
James

On Mon, Aug 20, 2012 at 2:45 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hello all,
>
> I need to refresh my list of people who have Web dev skills and are
> willing to assist with things like Web site design, programming, and
> maintenance. If you have any of the following skills, or know someone who
> does and who is willing to donate a few hours, please email me offlist:
>
> HTML5
> CSS2/3
> Javascript/JQuery
> Drupal/PHP
> Wordpress/PHP
> Django/Python
> Web design/graphics
> Apache administration in LAMP environment
>
> Please note that not every single skill is currently in demand; we are
> trying to get a feel for who knows what and hopefully anticipate future
> needs.
>
> Thanks!
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 at styleflare.com  Tue Aug 21 04:36:22 2012
From: david at styleflare.com (David J)
Date: Mon, 20 Aug 2012 20:36:22 -0400
Subject: [Freeswitch-users] FreeSWITCH Project Looking For Volunteers:
 Web Site Dev/Maint
In-Reply-To: <CALfx8k1i79u5zuhxyZy-QtZba5p9+zkk6fFLkw2RFy+K3K_uqQ@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
	<CALfx8k1i79u5zuhxyZy-QtZba5p9+zkk6fFLkw2RFy+K3K_uqQ@mail.gmail.com>
Message-ID: <CAPoPLj_cW2Q8V=FhGbAqRVYCRHhQx0hsQvP_T2DyO=CkVc7ffw@mail.gmail.com>

Congrats that's true double duty.
On Aug 20, 2012 8:35 PM, "James" <freeswitch at scottisheyes.com> wrote:

> I've got Django/Python, HTML, CSS, jQuery along with some server admin
> knowledge.  I'll help if I can, time dependent (I'm a new dad to twins, and
> yes, I like announcing it).
>
> Cheers,
> James
>
> On Mon, Aug 20, 2012 at 2:45 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Hello all,
>>
>> I need to refresh my list of people who have Web dev skills and are
>> willing to assist with things like Web site design, programming, and
>> maintenance. If you have any of the following skills, or know someone who
>> does and who is willing to donate a few hours, please email me offlist:
>>
>> HTML5
>> CSS2/3
>> Javascript/JQuery
>> Drupal/PHP
>> Wordpress/PHP
>> Django/Python
>> Web design/graphics
>> Apache administration in LAMP environment
>>
>> Please note that not every single skill is currently in demand; we are
>> trying to get a feel for who knows what and hopefully anticipate future
>> needs.
>>
>> Thanks!
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jmesquita at freeswitch.org  Tue Aug 21 04:40:18 2012
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 20 Aug 2012 21:40:18 -0300
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Project Looking
 For Volunteers: Web Site Dev/Maint
In-Reply-To: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
Message-ID: <CABH8QX0=noTUGUtJqKAQA4LxxVJizX3roPR3Jbhs4Pu7hg4g2g@mail.gmail.com>

MC, you know you can always count on me.

Python, JQuery, Javascript and lots of willing to learn new crap. :-)

Jo?o Mesquita



On Mon, Aug 20, 2012 at 6:45 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hello all,
>
> I need to refresh my list of people who have Web dev skills and are
> willing to assist with things like Web site design, programming, and
> maintenance. If you have any of the following skills, or know someone who
> does and who is willing to donate a few hours, please email me offlist:
>
> HTML5
> CSS2/3
> Javascript/JQuery
> Drupal/PHP
> Wordpress/PHP
> Django/Python
> Web design/graphics
> Apache administration in LAMP environment
>
> Please note that not every single skill is currently in demand; we are
> trying to get a feel for who knows what and hopefully anticipate future
> needs.
>
> Thanks!
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From mayamatakeshi at gmail.com  Tue Aug 21 04:53:17 2012
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 21 Aug 2012 09:53:17 +0900
Subject: [Freeswitch-users] Initiating application jitterbuffer multiple
	times
Message-ID: <CABaNFCZzw_jdEMUo3v6JfVbKO=9kKGdNLsKRmkEM8QZ963AL2g@mail.gmail.com>

Hello,
is it OK to start application jitterbuffer multiple times?
For example, before bridging a call I would do this:

  <action application="jitterbuffer" data="60:200:20"/>

then if later the call is transferred back to the dialplan, I would
call the above again before doing a new bridge.
Is there any chance this to be allocating extra resources?

Regards,
takeshi
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From gabe at gundy.org  Tue Aug 21 05:09:53 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 20 Aug 2012 19:09:53 -0600
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKzWOxXKCQnH+cDshF95FruwSFTxTyc-rXq8Ww81mXXbJe-2tA@mail.gmail.com>
References: <CAKzWOxXKCQnH+cDshF95FruwSFTxTyc-rXq8Ww81mXXbJe-2tA@mail.gmail.com>
Message-ID: <CAMwZ-t+Tgeh2RQDoPoWuR-22_cFJu1WGFtik8vWtXyH7TZiEPw@mail.gmail.com>

On Mon, Aug 20, 2012 at 11:56 AM, Michael Collins <marketing at cluecon.com> wrote:
> Lastly we'd like to let everyone know that the ClueCon videos will be made
> available in the coming weeks and months. Please give us some time to do a
> little editing before we release them all. It will be worth the wait!

I was there and I can't wait to see these :)  Did they capture video
of all the sessions?


Gabe


From roger.castaldo at gmail.com  Tue Aug 21 05:29:57 2012
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Mon, 20 Aug 2012 21:29:57 -0400
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Project Looking
 For Volunteers: Web Site Dev/Maint
In-Reply-To: <CABH8QX0=noTUGUtJqKAQA4LxxVJizX3roPR3Jbhs4Pu7hg4g2g@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
	<CABH8QX0=noTUGUtJqKAQA4LxxVJizX3roPR3Jbhs4Pu7hg4g2g@mail.gmail.com>
Message-ID: <CAAYf2LuSbtY4LEOuGCfAO6nCfcbHXqOTCmf3DrrGDowyemHjoA@mail.gmail.com>

I have all around Web experience as long as I have the time I an up for
helping...my python is rusty as is my php though
On Aug 20, 2012 7:41 PM, "Jo?o Mesquita" <jmesquita at freeswitch.org> wrote:

> MC, you know you can always count on me.
>
> Python, JQuery, Javascript and lots of willing to learn new crap. :-)
>
> Jo?o Mesquita
>
>
>
> On Mon, Aug 20, 2012 at 6:45 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Hello all,
>>
>> I need to refresh my list of people who have Web dev skills and are
>> willing to assist with things like Web site design, programming, and
>> maintenance. If you have any of the following skills, or know someone who
>> does and who is willing to donate a few hours, please email me offlist:
>>
>> HTML5
>> CSS2/3
>> Javascript/JQuery
>> Drupal/PHP
>> Wordpress/PHP
>> Django/Python
>> Web design/graphics
>> Apache administration in LAMP environment
>>
>> Please note that not every single skill is currently in demand; we are
>> trying to get a feel for who knows what and hopefully anticipate future
>> needs.
>>
>> Thanks!
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-list at puzzled.xs4all.nl  Tue Aug 21 06:21:15 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Tue, 21 Aug 2012 04:21:15 +0200
Subject: [Freeswitch-users] mod_portaudio cannot find input or output
 device
In-Reply-To: <06fb01cd7f2f$75b2b950$61182bf0$@launch3.net>
References: <021301cd7c72$6adff2f0$409fd8d0$@launch3.net>	<502E3CE7.4070902@puzzled.xs4all.nl>	<01e001cd7c9c$3c454f10$b4cfed30$@launch3.net>	<CALn233r1Y0QfHzkT7fU_N8neSztcMLhx38UR2ZnjNy_Xydiejg@mail.gmail.com>	<020301cd7cb8$d5985790$80c906b0$@launch3.net>	<00ea01cd7f0a$b9474a70$2bd5df50$@launch3.net>	<5032976B.7050208@puzzled.xs4all.nl>	<011501cd7f15$8d9315c0$a8b94140$@launch3.net>
	<CALn233qreDcwqDrB20EKDZK1nHCo2SeiO4U68evhK_7WZu=hsA@mail.gmail.com>
	<06fb01cd7f2f$75b2b950$61182bf0$@launch3.net>
Message-ID: <5032F09B.4060806@puzzled.xs4all.nl>

On 21-08-12 01:56, Brett Wilson wrote:
> How do I go about debugging it if there is no crash? I looked at the wiki
> but looks like gdb would only work if there is a crash.

I guess it would boil down to adding debug statements in the code.

On a different note, did you try different versions of the portaudio 
libs? Looking at http://www.portaudio.com/download.html there seem to be 
several versions that you could try. Maybe try to match the date of the 
mod_portaudio commits to the stable release that comes closest to that date.

Regards,
Patrick





From cmrienzo at gmail.com  Tue Aug 21 06:24:31 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Mon, 20 Aug 2012 22:24:31 -0400
Subject: [Freeswitch-users] Initiating application jitterbuffer multiple
	times
In-Reply-To: <CABaNFCZzw_jdEMUo3v6JfVbKO=9kKGdNLsKRmkEM8QZ963AL2g@mail.gmail.com>
References: <CABaNFCZzw_jdEMUo3v6JfVbKO=9kKGdNLsKRmkEM8QZ963AL2g@mail.gmail.com>
Message-ID: <CAD-qRkhuEBUuD+JBpJ4K1osbdEfvoOUCbg56vQYNgWf9SXyKxQ@mail.gmail.com>

When you execute the jitterbuffer application, it will start the jitter
buffer.  The jitter buffer is paused during bridge by default and is
resumed when bridge ends.  You can set sip_jitter_buffer_during_bridge=true
if you want the jitter buffer to keep running during bridge.

It's ok to call the jitterbuffer application multiple times, but what you
described didn't make sense to me.  When do you want the jitter buffer on?
 When do you want it paused?

Chris


On Mon, Aug 20, 2012 at 8:53 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> Hello,
> is it OK to start application jitterbuffer multiple times?
> For example, before bridging a call I would do this:
>
>   <action application="jitterbuffer" data="60:200:20"/>
>
> then if later the call is transferred back to the dialplan, I would call the above again before doing a new bridge.
> Is there any chance this to be allocating extra resources?
>
> Regards,
> takeshi
>
>
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From jason at jasonjgw.net  Tue Aug 21 06:49:33 2012
From: jason at jasonjgw.net (Jason White)
Date: Tue, 21 Aug 2012 02:49:33 +0000 (UTC)
Subject: [Freeswitch-users] FreeSWITCH and Pulse Audio
Message-ID: <k0usvt$h1r$1@ger.gmane.org>

What, if any, is the right way to set up FreeSWITCH as a soft-phone on a
system with Pulse Audio handling the audio device?

I found a bug on this topic that was never fixed, so I'm wondering what the
solution now is and in which direction development is heading.

In the admittedly old version of FreeSWITCH that I'm running, mod_portaudio
fails to find the device which is being handled by Pulse Audio as the default
(it simply doesn't appear in the output of pa devlist).

I know there is also an Alsa module for FreeSWITCH, which used to be rather
limited in functionality and not a substitute for mod_portaudio.

Thus my basic question is where FreeSWITCH is heading with regard to audio I/O
under Linux. Is there anything interesting in the pipeline?




From mayamatakeshi at gmail.com  Tue Aug 21 06:52:56 2012
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 21 Aug 2012 11:52:56 +0900
Subject: [Freeswitch-users] Initiating application jitterbuffer multiple
	times
In-Reply-To: <CAD-qRkhuEBUuD+JBpJ4K1osbdEfvoOUCbg56vQYNgWf9SXyKxQ@mail.gmail.com>
References: <CABaNFCZzw_jdEMUo3v6JfVbKO=9kKGdNLsKRmkEM8QZ963AL2g@mail.gmail.com>
	<CAD-qRkhuEBUuD+JBpJ4K1osbdEfvoOUCbg56vQYNgWf9SXyKxQ@mail.gmail.com>
Message-ID: <CABaNFCZ5ymSALKMPztxZYxFWQM03thh-6MJubpfnQ6HfCQnhqw@mail.gmail.com>

On Tue, Aug 21, 2012 at 11:24 AM, Christopher Rienzo <cmrienzo at gmail.com>wrote:

> When you execute the jitterbuffer application, it will start the jitter
> buffer.  The jitter buffer is paused during bridge by default and is
> resumed when bridge ends.
>

OK. Thanks.
But this is a bit surprising. What would be the rationale for this?
I mean, at least in the company I work for, jitterbuffer is used to solve
problems during SIP calls. We are not so far trying to solve any problem
with unbridged calls (like calls to voicemail) because we don't have them
yet.


>  You can set sip_jitter_buffer_during_bridge=true if you want the jitter
> buffer to keep running during bridge.
>

OK. Noted.


>
> It's ok to call the jitterbuffer application multiple times,
>

OK.


> but what you described didn't make sense to me.
>

Well, my doubt was if there is any problem of calling the jitterbuffer app
multiple times. This will happen in case the channel is sent back to the
dialplan (blind transfer) and if I don't check if the jitterbuffer was
already set or not.
I asked this because I saw things in the past like calling start_dtmf more
than once causing duplication of DTMF notification (meaning more than one
resource was allocated for the task). So I would just want to confirm this
will not be the case for jitterbuffer. Otherwise, I would have to check if
jitterbuffer was already called and avoid calling it again (nothing
complex, but I will not add extra code if there is not need).
(I would set it in the sip profile, but as the wiki says, there you cannot
set the jitter buffer max length)


>  When do you want the jitter buffer on?
>

Actually I want it enabled all the time no matter if the channel is bridged
or not.


>  When do you want it paused?
>

Never.

Regards,
takeshi


>
>
> On Mon, Aug 20, 2012 at 8:53 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:
>
>> Hello,
>> is it OK to start application jitterbuffer multiple times?
>> For example, before bridging a call I would do this:
>>
>>   <action application="jitterbuffer" data="60:200:20"/>
>>
>>
>> then if later the call is transferred back to the dialplan, I would call the above again before doing a new bridge.
>> Is there any chance this to be allocating extra resources?
>>
>> Regards,
>> takeshi
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Tue Aug 21 09:25:55 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 21 Aug 2012 00:25:55 -0500
Subject: [Freeswitch-users] Initiating application jitterbuffer multiple
	times
In-Reply-To: <CABaNFCZ5ymSALKMPztxZYxFWQM03thh-6MJubpfnQ6HfCQnhqw@mail.gmail.com>
References: <CABaNFCZzw_jdEMUo3v6JfVbKO=9kKGdNLsKRmkEM8QZ963AL2g@mail.gmail.com>
	<CAD-qRkhuEBUuD+JBpJ4K1osbdEfvoOUCbg56vQYNgWf9SXyKxQ@mail.gmail.com>
	<CABaNFCZ5ymSALKMPztxZYxFWQM03thh-6MJubpfnQ6HfCQnhqw@mail.gmail.com>
Message-ID: <CAKbxfG-Mo+7wdtYzpODH6K3Sh=V-SnyqRWurkw72zTjAvpQK+g@mail.gmail.com>

If the jitterbuffer is already on, calling it again will just resize
it, so setting it to the same is redundant but harmless.

If you are surprised by why the jitterbuffer is paused during bridge:

If both sides of a bridge are RTP and both sides have a jb, its fairly
useless.  In fact if anything, it can worsen call quality.

You should only run jitterbuffers at points of termination change of
protocol.  Examples, if FS was hosting a conference or IVR, if you are
bridging the call to a phone for instance, you want to not use a
jitterbuffer because you want to preserve the original timestamps so
your phone can use its own jitterbuffer.

For special examples where you are using FS jitterbuffer in front of
something else that may not have one or some other special
circumstance you can use the setting chris mentioned to leave it
running.



On Mon, Aug 20, 2012 at 9:52 PM, mayamatakeshi <mayamatakeshi at gmail.com> wrote:
>
> On Tue, Aug 21, 2012 at 11:24 AM, Christopher Rienzo <cmrienzo at gmail.com>
> wrote:
>>
>> When you execute the jitterbuffer application, it will start the jitter
>> buffer.  The jitter buffer is paused during bridge by default and is resumed
>> when bridge ends.
>
>
> OK. Thanks.
> But this is a bit surprising. What would be the rationale for this?
> I mean, at least in the company I work for, jitterbuffer is used to solve
> problems during SIP calls. We are not so far trying to solve any problem
> with unbridged calls (like calls to voicemail) because we don't have them
> yet.
>
>>
>>  You can set sip_jitter_buffer_during_bridge=true if you want the jitter
>> buffer to keep running during bridge.
>
>
> OK. Noted.
>
>>
>>
>> It's ok to call the jitterbuffer application multiple times,
>
>
> OK.
>
>>
>> but what you described didn't make sense to me.
>
>
> Well, my doubt was if there is any problem of calling the jitterbuffer app
> multiple times. This will happen in case the channel is sent back to the
> dialplan (blind transfer) and if I don't check if the jitterbuffer was
> already set or not.
> I asked this because I saw things in the past like calling start_dtmf more
> than once causing duplication of DTMF notification (meaning more than one
> resource was allocated for the task). So I would just want to confirm this
> will not be the case for jitterbuffer. Otherwise, I would have to check if
> jitterbuffer was already called and avoid calling it again (nothing complex,
> but I will not add extra code if there is not need).
> (I would set it in the sip profile, but as the wiki says, there you cannot
> set the jitter buffer max length)
>
>>
>>  When do you want the jitter buffer on?
>
>
> Actually I want it enabled all the time no matter if the channel is bridged
> or not.
>
>>
>>  When do you want it paused?
>
>
> Never.
>
> Regards,
> takeshi
>
>>
>>
>>
>> On Mon, Aug 20, 2012 at 8:53 PM, mayamatakeshi <mayamatakeshi at gmail.com>
>> wrote:
>>>
>>> Hello,
>>> is it OK to start application jitterbuffer multiple times?
>>> For example, before bridging a call I would do this:
>>>
>>>   <action application="jitterbuffer" data="60:200:20"/>
>>>
>>>
>>>
>>>
>>> then if later the call is transferred back to the dialplan, I would call
>>> the above again before doing a new bridge.
>>> Is there any chance this to be allocating extra resources?
>>>
>>> Regards,
>>> takeshi
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From mayamatakeshi at gmail.com  Tue Aug 21 09:48:32 2012
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 21 Aug 2012 14:48:32 +0900
Subject: [Freeswitch-users] Initiating application jitterbuffer multiple
	times
In-Reply-To: <CAKbxfG-Mo+7wdtYzpODH6K3Sh=V-SnyqRWurkw72zTjAvpQK+g@mail.gmail.com>
References: <CABaNFCZzw_jdEMUo3v6JfVbKO=9kKGdNLsKRmkEM8QZ963AL2g@mail.gmail.com>
	<CAD-qRkhuEBUuD+JBpJ4K1osbdEfvoOUCbg56vQYNgWf9SXyKxQ@mail.gmail.com>
	<CABaNFCZ5ymSALKMPztxZYxFWQM03thh-6MJubpfnQ6HfCQnhqw@mail.gmail.com>
	<CAKbxfG-Mo+7wdtYzpODH6K3Sh=V-SnyqRWurkw72zTjAvpQK+g@mail.gmail.com>
Message-ID: <CABaNFCandJhvHUM9ze8Yk2n5ZtvD0k3PnVXm_vm-8ExJvJHNpQ@mail.gmail.com>

On Tue, Aug 21, 2012 at 2:25 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> If the jitterbuffer is already on, calling it again will just resize
> it, so setting it to the same is redundant but harmless.
>
> If you are surprised by why the jitterbuffer is paused during bridge:
>
> If both sides of a bridge are RTP and both sides have a jb, its fairly
> useless.  In fact if anything, it can worsen call quality.
>
> You should only run jitterbuffers at points of termination change of
> protocol.  Examples, if FS was hosting a conference or IVR, if you are
> bridging the call to a phone for instance, you want to not use a
> jitterbuffer because you want to preserve the original timestamps so
> your phone can use its own jitterbuffer.
>

Ah. OK.
But I failed to mention that I am using FS for transcoding.
Our rationale was this:
  - we have terminals with bad jitter and they can select from a variety of
codecs.
  - FS needs jitter buffers in both sides of the bridge to properly
recompose the audio and do the transcoding.
Is this OK?
If yes, I understand things will not be that simple because then I need to
check if transcoding is needed or not and allow for jitterbuffers to only
be enabled in this case.

Regards,
takeshi



>
> For special examples where you are using FS jitterbuffer in front of
> something else that may not have one or some other special
> circumstance you can use the setting chris mentioned to leave it
> running.
>
>
>
> On Mon, Aug 20, 2012 at 9:52 PM, mayamatakeshi <mayamatakeshi at gmail.com>
> wrote:
> >
> > On Tue, Aug 21, 2012 at 11:24 AM, Christopher Rienzo <cmrienzo at gmail.com
> >
> > wrote:
> >>
> >> When you execute the jitterbuffer application, it will start the jitter
> >> buffer.  The jitter buffer is paused during bridge by default and is
> resumed
> >> when bridge ends.
> >
> >
> > OK. Thanks.
> > But this is a bit surprising. What would be the rationale for this?
> > I mean, at least in the company I work for, jitterbuffer is used to solve
> > problems during SIP calls. We are not so far trying to solve any problem
> > with unbridged calls (like calls to voicemail) because we don't have them
> > yet.
> >
> >>
> >>  You can set sip_jitter_buffer_during_bridge=true if you want the jitter
> >> buffer to keep running during bridge.
> >
> >
> > OK. Noted.
> >
> >>
> >>
> >> It's ok to call the jitterbuffer application multiple times,
> >
> >
> > OK.
> >
> >>
> >> but what you described didn't make sense to me.
> >
> >
> > Well, my doubt was if there is any problem of calling the jitterbuffer
> app
> > multiple times. This will happen in case the channel is sent back to the
> > dialplan (blind transfer) and if I don't check if the jitterbuffer was
> > already set or not.
> > I asked this because I saw things in the past like calling start_dtmf
> more
> > than once causing duplication of DTMF notification (meaning more than one
> > resource was allocated for the task). So I would just want to confirm
> this
> > will not be the case for jitterbuffer. Otherwise, I would have to check
> if
> > jitterbuffer was already called and avoid calling it again (nothing
> complex,
> > but I will not add extra code if there is not need).
> > (I would set it in the sip profile, but as the wiki says, there you
> cannot
> > set the jitter buffer max length)
> >
> >>
> >>  When do you want the jitter buffer on?
> >
> >
> > Actually I want it enabled all the time no matter if the channel is
> bridged
> > or not.
> >
> >>
> >>  When do you want it paused?
> >
> >
> > Never.
> >
> > Regards,
> > takeshi
> >
> >>
> >>
> >>
> >> On Mon, Aug 20, 2012 at 8:53 PM, mayamatakeshi <mayamatakeshi at gmail.com
> >
> >> wrote:
> >>>
> >>> Hello,
> >>> is it OK to start application jitterbuffer multiple times?
> >>> For example, before bridging a call I would do this:
> >>>
> >>>   <action application="jitterbuffer" data="60:200:20"/>
> >>>
> >>>
> >>>
> >>>
> >>> then if later the call is transferred back to the dialplan, I would
> call
> >>> the above again before doing a new bridge.
> >>> Is there any chance this to be allocating extra resources?
> >>>
> >>> Regards,
> >>> takeshi
> >>>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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
> >>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 obiebrown at gmail.com  Tue Aug 21 09:50:58 2012
From: obiebrown at gmail.com (Obie Brown)
Date: Tue, 21 Aug 2012 15:20:58 +0930
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKzWOxXKCQnH+cDshF95FruwSFTxTyc-rXq8Ww81mXXbJe-2tA@mail.gmail.com>
References: <CAKzWOxXKCQnH+cDshF95FruwSFTxTyc-rXq8Ww81mXXbJe-2tA@mail.gmail.com>
Message-ID: <CA+aeUL9LfaX2F_JCLgkKPdnY3UEGXYu4W2nnwx353YDrvDbO1A@mail.gmail.com>

I would be very interested to work with someone or a team to produce a
Vestec ASR (automated speech recognition) app.

What I would like to build is a ASR app for Voice Broadcasting. I am very
new to FreeSWITCH but would have some money to put behind this app if
someone or a team was interested in working with me.

If you are interested contact me and we can talk more.


Thanks,
Obie Brown
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From gabe at gundy.org  Tue Aug 21 10:32:14 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 21 Aug 2012 00:32:14 -0600
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Project Looking
 For Volunteers: Web Site Dev/Maint
In-Reply-To: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
Message-ID: <CAMwZ-t+6xdX6wwwE7vUmvaHS9D50FN5o5iR4Qz3y+MvYzNr-Pw@mail.gmail.com>

On Mon, Aug 20, 2012 at 3:45 PM, Michael Collins <msc at freeswitch.org> wrote:
> HTML5
> CSS2/3
> Javascript/JQuery
> Django/Python
> Web design/graphics
> Apache administration in LAMP environment

The folks at Izeni are happy to help where we can (with the trimmed
list of techs).  We feel it's one of the ways we can give back to the
FreeSWITCH project and community that we benefit so much from.

We're look forward to working with the others who join in.


Best,
Gabe


From vinolasbcn at gmail.com  Tue Aug 21 03:41:41 2012
From: vinolasbcn at gmail.com (=?ISO-8859-1?Q?Josep_Maria_Vi=F1olas?=)
Date: Tue, 21 Aug 2012 01:41:41 +0200
Subject: [Freeswitch-users] mod_gsmopen: ROAMING Network ERROR
Message-ID: <CAD=Wdqi_rbz3Z0XrYcS_CP29wSV94G8R7cJpHKRqZoeNorhmxw@mail.gmail.com>

I have a Huawei E169 with a SIM from a "virtual provider" that uses other
provider's networks, so it is allways technically 'roaming'.

With mod_gsmopen the problem is that throws an error about "ROAMING
Network" and also an ALERT. I don't know if that is why it is not able to
make calls. In that case, how can I tell the module that roaming is not a
bad thing??

This is freeswitch debug output when I load mod_gsmmobile:
------------------------------------------
2012-08-21 01:35:49.104471 [CONSOLE] switch_loadable_module.c:1328
Successfully Loaded [mod_gsmopen]
2012-08-21 01:35:49.104471 [NOTICE] switch_loadable_module.c:146 Adding
Endpoint 'gsmopen'
2012-08-21 01:35:49.126116 [NOTICE] switch_loadable_module.c:298 Adding API
Function 'gsm'
2012-08-21 01:35:49.146636 [NOTICE] switch_loadable_module.c:298 Adding API
Function 'gsmopen'
2012-08-21 01:35:49.164548 [NOTICE] switch_loadable_module.c:298 Adding API
Function 'gsmopen_boost_audio'
2012-08-21 01:35:49.164548 [NOTICE] switch_loadable_module.c:298 Adding API
Function 'gsmopen_dump'
2012-08-21 01:35:49.185691 [NOTICE] switch_loadable_module.c:298 Adding API
Function 'gsmopen_sendsms'
2012-08-21 01:35:49.185691 [NOTICE] switch_loadable_module.c:403 Adding
Chat interface 'sms'

+OK Reloading XML
+OK

freeswitch at raspberrypi> 2012-08-21 01:35:49.344554 [ERR]
gsmopen_protocol.cpp:1027 rev ccae5cd|07bc7ba[(nil)|37     ][ERRORA  1027
][gsmmodem  ][-1, 0, 0] |+CREG: 1,5| CELLPHONE is registered to a ROAMING
network
2012-08-21 01:35:49.344554 [ERR] mod_gsmopen.cpp:2928 rev
ccae5cd|07bc7ba[(nil)|37     ][ERRORA  2928 ][gsmmodem  ][-1, 0, 0] ALARM
on interface gsmmodem:
------------------------------------------

And this when I try to make a call through usb dongle:
------------------------------------------
2012-08-21 01:38:21.504633 [NOTICE] switch_channel.c:946 New Channel
sofia/internal/1000 at 192.168.1.221 [208677de-eb20-11e1-bb15-eb03c1e5a0a5]
2012-08-21 01:38:21.584605 [INFO] mod_dialplan_xml.c:485 Processing 1000
<1000>->9876 in context default
2012-08-21 01:38:21.684731 [NOTICE] switch_channel.c:946 New Channel
gsmopen/gsmmodem/653379498 [20a13696-eb20-11e1-bb1b-eb03c1e5a0a5]
2012-08-21 01:38:21.744655 [NOTICE] mod_gsmopen.cpp:2176 Pre-Answer
gsmopen/gsmmodem/653379498!
2012-08-21 01:38:21.924971 [ERR] gsmopen_protocol.cpp:2340 rev
ccae5cd|07bc7ba[(nil)|37     ][ERRORA  2340 ][gsmmodem  ][-1,24, 3] dial
command failed, dial string was: ATD65337XXXX;
2012-08-21 01:38:22.044648 [INFO] switch_ivr_originate.c:3309 Sending early
media
2012-08-21 01:38:22.304765 [NOTICE] sofia_glue.c:4176 Pre-Answer
sofia/internal/1000 at 192.168.1.221!
2012-08-21 01:38:31.684677 [NOTICE] sofia.c:6836 Hangup sofia/internal/
1000 at 192.168.1.221 [CS_EXECUTE] [ORIGINATOR_CANCEL]
2012-08-21 01:38:31.724776 [NOTICE] switch_ivr_bridge.c:674 Hangup
gsmopen/gsmmodem/653379498 [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL]
2012-08-21 01:38:32.345425 [NOTICE] switch_core_session.c:1446 Session 1
(sofia/internal/1000 at 192.168.1.221) Ended
2012-08-21 01:38:32.365588 [NOTICE] switch_core_session.c:1450 Close
Channel sofia/internal/1000 at 192.168.1.221 [CS_DESTROY]
2012-08-21 01:38:32.424632 [NOTICE] switch_core_session.c:1446 Session 2
(gsmopen/gsmmodem/653379498) Ended
2012-08-21 01:38:32.444653 [NOTICE] switch_core_session.c:1450 Close
Channel gsmopen/gsmmodem/653379498 [CS_DESTROY]
------------------------------------------
** From extension 1000 I hear a calling tone even the Freeswitch log says
that ATD+N? command failed.

Anyone knows if the problem is the ROAMING Network error??
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From dvl36.ripe.nick at gmail.com  Tue Aug 21 11:07:02 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Tue, 21 Aug 2012 10:07:02 +0300
Subject: [Freeswitch-users] mod_gsmopen: ROAMING Network ERROR
In-Reply-To: <CAD=Wdqi_rbz3Z0XrYcS_CP29wSV94G8R7cJpHKRqZoeNorhmxw@mail.gmail.com>
References: <CAD=Wdqi_rbz3Z0XrYcS_CP29wSV94G8R7cJpHKRqZoeNorhmxw@mail.gmail.com>
Message-ID: <CAMjaK3B-Y+nswvVnj49Jh_+sZCJKvcXiSmhJ7ksMWo6UXbGt9Q@mail.gmail.com>

Hello.
I have the same "ROAMING Network" ALERTs, it's annoying, but everything
working OK.
Checked with  Huawei E1550.

P.S. Are you sure that your Huawei E169  is voice enabled?

2012/8/21 Josep Maria Vi?olas <vinolasbcn at gmail.com>

> I have a Huawei E169 with a SIM from a "virtual provider" that uses other
> provider's networks, so it is allways technically 'roaming'.
>
> With mod_gsmopen the problem is that throws an error about "ROAMING
> Network" and also an ALERT. I don't know if that is why it is not able to
> make calls. In that case, how can I tell the module that roaming is not a
> bad thing??
>
> This is freeswitch debug output when I load mod_gsmmobile:
> ------------------------------------------
>
...

> ------------------------------------------
> ** From extension 1000 I hear a calling tone even the Freeswitch log says
> that ATD+N? command failed.
>
> Anyone knows if the problem is the ROAMING Network error??
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eagle.antonio at gmail.com  Tue Aug 21 11:56:44 2012
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 21 Aug 2012 07:56:44 +0000
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Project Looking
 For Volunteers: Web Site Dev/Maint
In-Reply-To: <CAMwZ-t+6xdX6wwwE7vUmvaHS9D50FN5o5iR4Qz3y+MvYzNr-Pw@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
	<CAMwZ-t+6xdX6wwwE7vUmvaHS9D50FN5o5iR4Qz3y+MvYzNr-Pw@mail.gmail.com>
Message-ID: <CAF+3j-tMcwmYkUQQ1RV=Mo72d6BU2gYDyU7BT5P3eKe_9OYT8A@mail.gmail.com>

Hello guys
Regarding the Django / Python / Apache .

Count me in :)

Regards
Ant?nio Teixeira

2012/8/21 Gabriel Gunderson <gabe at gundy.org>

> On Mon, Aug 20, 2012 at 3:45 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > HTML5
> > CSS2/3
> > Javascript/JQuery
> > Django/Python
> > Web design/graphics
> > Apache administration in LAMP environment
>
> The folks at Izeni are happy to help where we can (with the trimmed
> list of techs).  We feel it's one of the ways we can give back to the
> FreeSWITCH project and community that we benefit so much from.
>
> We're look forward to working with the others who join in.
>
>
> Best,
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sameer2k3t at gmail.com  Tue Aug 21 12:09:52 2012
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Tue, 21 Aug 2012 13:09:52 +0500
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
Message-ID: <CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>

It didn't for me, according to wvdial my modem is ttyUSB0 and ttyUSB3

On Mon, Aug 20, 2012 at 1:43 PM, Dmitry Lysenko
<dvl36.ripe.nick at gmail.com>wrote:

>
> Hello.
> Try:
>  <param name="controldevice_name" value="/dev/ttyUSB2"/>
>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
> It's worked for me.
>
> 2012/8/20 Sameer Khan <sameer2k3t at gmail.com>
>
>> Hi,
>>
>> I can't get audio working with e1550
>>
>> I have this config :
>>     <interface id="1" name="gsm01">
>>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>       <param name="controldevice_audio_name" value="/dev/ttyUSB3"/>
>>
>> I tried swaping 0 and 3 but having same results
>>
>> I am originating a call from softphone and answering it on cell phone but
>> softphone doesn't get notified of being answered and keeps ringing
>>
>> Please help
>>
>> Thanks
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 andrew at cassidywebservices.co.uk  Tue Aug 21 13:13:39 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Tue, 21 Aug 2012 10:13:39 +0100
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Project Looking
 For Volunteers: Web Site Dev/Maint
In-Reply-To: <CAF+3j-tMcwmYkUQQ1RV=Mo72d6BU2gYDyU7BT5P3eKe_9OYT8A@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
	<CAMwZ-t+6xdX6wwwE7vUmvaHS9D50FN5o5iR4Qz3y+MvYzNr-Pw@mail.gmail.com>
	<CAF+3j-tMcwmYkUQQ1RV=Mo72d6BU2gYDyU7BT5P3eKe_9OYT8A@mail.gmail.com>
Message-ID: <CAMbW9o+94bnu6ntg4nv+jYVgxOGvk8UmisMm6e0SBVZTKNYmuA@mail.gmail.com>

Same as Gabe, except my design abilities are shockingly bad.

On 21 August 2012 08:56, Antonio Teixeira <eagle.antonio at gmail.com> wrote:

> Hello guys
> Regarding the Django / Python / Apache .
>
> Count me in :)
>
> Regards
> Ant?nio Teixeira
>
>
> 2012/8/21 Gabriel Gunderson <gabe at gundy.org>
>
>> On Mon, Aug 20, 2012 at 3:45 PM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > HTML5
>> > CSS2/3
>> > Javascript/JQuery
>> > Django/Python
>> > Web design/graphics
>> > Apache administration in LAMP environment
>>
>> The folks at Izeni are happy to help where we can (with the trimmed
>> list of techs).  We feel it's one of the ways we can give back to the
>> FreeSWITCH project and community that we benefit so much from.
>>
>> We're look forward to working with the others who join in.
>>
>>
>> Best,
>> Gabe
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From dvl36.ripe.nick at gmail.com  Tue Aug 21 13:21:30 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Tue, 21 Aug 2012 12:21:30 +0300
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
Message-ID: <CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>

Are you using "option" linux driver?
I checked Huawei E1550, E153 and E173 modems on some systems and they have
only 3 virtual serial ports.(using "option" driver)
Also you can check manually with minicom.
Usually ttyUSB0 is accepting  AT-commands, but will not reply with OK or
ERROR, ttyUSB1 is used for audio, and ttyUSB2 is accepting AT-commands and
replying  as normal serial modem.


2012/8/21 Sameer Khan <sameer2k3t at gmail.com>

> It didn't for me, according to wvdial my modem is ttyUSB0 and ttyUSB3
>
> On Mon, Aug 20, 2012 at 1:43 PM, Dmitry Lysenko <dvl36.ripe.nick at gmail.com
> > wrote:
>
>>
>> Hello.
>> Try:
>>  <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>> It's worked for me.
>>
>> 2012/8/20 Sameer Khan <sameer2k3t at gmail.com>
>>
>>>  Hi,
>>>
>>> I can't get audio working with e1550
>>>
>>> I have this config :
>>>     <interface id="1" name="gsm01">
>>>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB3"/>
>>>
>>> I tried swaping 0 and 3 but having same results
>>>
>>> I am originating a call from softphone and answering it on cell phone
>>> but softphone doesn't get notified of being answered and keeps ringing
>>>
>>> Please help
>>>
>>> Thanks
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sameer2k3t at gmail.com  Tue Aug 21 13:35:50 2012
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Tue, 21 Aug 2012 14:35:50 +0500
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
	<CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
Message-ID: <CAA2gtHBXf-RfZ4eFB1k7NfCaYccqU6uK78NnxxBL0C=dO2PPBA@mail.gmail.com>

I guess so, I have CentOS 6 minimal installed

how can I identify if this is option driver ?

Yes I saw this message on ttyUSB0

I can not load gsmopen with ttyUSB2 or ttyUSB1

On Tue, Aug 21, 2012 at 2:21 PM, Dmitry Lysenko
<dvl36.ripe.nick at gmail.com>wrote:

> Are you using "option" linux driver?
> I checked Huawei E1550, E153 and E173 modems on some systems and they have
> only 3 virtual serial ports.(using "option" driver)
> Also you can check manually with minicom.
> Usually ttyUSB0 is accepting  AT-commands, but will not reply with OK or
> ERROR, ttyUSB1 is used for audio, and ttyUSB2 is accepting AT-commands and
> replying  as normal serial modem.
>
>
> 2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>
>> It didn't for me, according to wvdial my modem is ttyUSB0 and ttyUSB3
>>
>> On Mon, Aug 20, 2012 at 1:43 PM, Dmitry Lysenko <
>> dvl36.ripe.nick at gmail.com> wrote:
>>
>>>
>>> Hello.
>>> Try:
>>>  <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>> It's worked for me.
>>>
>>> 2012/8/20 Sameer Khan <sameer2k3t at gmail.com>
>>>
>>>>  Hi,
>>>>
>>>> I can't get audio working with e1550
>>>>
>>>> I have this config :
>>>>     <interface id="1" name="gsm01">
>>>>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB3"/>
>>>>
>>>> I tried swaping 0 and 3 but having same results
>>>>
>>>> I am originating a call from softphone and answering it on cell phone
>>>> but softphone doesn't get notified of being answered and keeps ringing
>>>>
>>>> Please help
>>>>
>>>> Thanks
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 dvl36.ripe.nick at gmail.com  Tue Aug 21 13:55:50 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Tue, 21 Aug 2012 12:55:50 +0300
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAA2gtHBXf-RfZ4eFB1k7NfCaYccqU6uK78NnxxBL0C=dO2PPBA@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
	<CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
	<CAA2gtHBXf-RfZ4eFB1k7NfCaYccqU6uK78NnxxBL0C=dO2PPBA@mail.gmail.com>
Message-ID: <CAMjaK3DCzKEi7Gep9eZLhGs4FGix3edCDCxtymJXCwWHP9FSjA@mail.gmail.com>

As root: lsmod | grep option
Should be something like that:
---
root at debian:~# lsmod | grep option
option                 15351  6
usb_wwan                8362  1 option
usbserial              26535  14 usb_wwan,option
usbcore               122921  10
ehci_hcd,usb_storage,usbserial,gspca_main,usb_wwan,option,gspca_zc3xx,snd_usbmidi_lib,snd_usb_audio
---
2012/8/21 Sameer Khan <sameer2k3t at gmail.com>

> I guess so, I have CentOS 6 minimal installed
>
> how can I identify if this is option driver ?
>
> Yes I saw this message on ttyUSB0
>
> I can not load gsmopen with ttyUSB2 or ttyUSB1
>
>
> On Tue, Aug 21, 2012 at 2:21 PM, Dmitry Lysenko <dvl36.ripe.nick at gmail.com
> > wrote:
>
>> Are you using "option" linux driver?
>> I checked Huawei E1550, E153 and E173 modems on some systems and they
>> have only 3 virtual serial ports.(using "option" driver)
>> Also you can check manually with minicom.
>> Usually ttyUSB0 is accepting  AT-commands, but will not reply with OK or
>> ERROR, ttyUSB1 is used for audio, and ttyUSB2 is accepting AT-commands and
>> replying  as normal serial modem.
>>
>>
>> 2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>
>>> It didn't for me, according to wvdial my modem is ttyUSB0 and ttyUSB3
>>>
>>> On Mon, Aug 20, 2012 at 1:43 PM, Dmitry Lysenko <
>>> dvl36.ripe.nick at gmail.com> wrote:
>>>
>>>>
>>>> Hello.
>>>> Try:
>>>>  <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>> It's worked for me.
>>>>
>>>> 2012/8/20 Sameer Khan <sameer2k3t at gmail.com>
>>>>
>>>>>  Hi,
>>>>>
>>>>> I can't get audio working with e1550
>>>>>
>>>>> I have this config :
>>>>>     <interface id="1" name="gsm01">
>>>>>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB3"/>
>>>>>
>>>>> I tried swaping 0 and 3 but having same results
>>>>>
>>>>> I am originating a call from softphone and answering it on cell phone
>>>>> but softphone doesn't get notified of being answered and keeps ringing
>>>>>
>>>>> Please help
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mi.ke at null.net  Tue Aug 21 13:58:24 2012
From: mi.ke at null.net (Mi Ke)
Date: Tue, 21 Aug 2012 05:58:24 -0400
Subject: [Freeswitch-users] mod_gsmopen serial init failed
Message-ID: <20120821095824.210970@gmx.com>

Hi All,

 I have problems getting mod_gsmopen to see my E1550. Both minicom and wvdial see the dongle on /dev/ttyUSB0:

 ttyUSB0<*1>: ATQ0 V1 E1 -- OK
 ttyUSB0<*1>: ATQ0 V1 E1 Z -- OK
 ttyUSB0<*1>: ATQ0 V1 E1 S0=0 -- OK
 ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 -- OK
 ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 -- OK
 ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0 -- OK
 ttyUSB0<*1>: Modem Identifier: ATI -- Manufacturer: huawei
 ttyUSB0<*1>: Speed 9600: AT -- OK
 ttyUSB0<*1>: Max speed is 9600; that should be safe.
 ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0 -- OK

 Welcome to minicom 2.3

 OPTIONS: I18n
 Compiled on Aug 19 2010, 05:50:19.
 Port /dev/ttyUSB0

 Press CTRL-A Z for help on special keys
 AT S7=45 S0=0 L1 V1 X4 &c1 E1 Q0
 OK 

 at+cpin?
 +CPIN: READY
 OK

 but mod_gsmopen does not want to see it:

 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1203 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1203 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 0 ?????
 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1204 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1204 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 1 ???^?
 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1205 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1205 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 2 ???????
 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1206 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1206 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 3 ??????
 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1207 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1207 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 4 ???
 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1208 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1208 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 5 ??
 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1209 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1209 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 6 ??
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1226 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1226 ][none ][-1,-1,-1] globals.debug=0
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1228 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1228 ][none ][-1,-1,-1] globals.debug=8
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1234 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1234 ][none ][-1,-1,-1] globals.dialplan=XML
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1240 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1240 ][none ][-1,-1,-1] globals.context=external
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1231 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1231 ][none ][-1,-1,-1] globals.hold_music=
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1237 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1237 ][none ][-1,-1,-1] globals.destination=5000
 2012-08-21 09:18:18.973501 [WARNING] mod_gsmopen.cpp:1653 rev ccae5cd|07bc7ba[(nil)|37 ][WARNINGA 1653 ][gsm00 ][-1, 0, 0] STARTING interface_id=1
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1654 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1654 ][gsm00 ][-1, 0, 0] id=1
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1655 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1655 ][gsm00 ][-1, 0, 0] name=gsm00
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1656 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1656 ][gsm00 ][-1, 0, 0] hold-music=
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1657 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1657 ][gsm00 ][-1, 0, 0] context=default
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1658 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1658 ][gsm00 ][-1, 0, 0] dialplan=XML
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1659 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1659 ][gsm00 ][-1, 0, 0] destination=5000
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1660 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1660 ][gsm00 ][-1, 0, 0] controldevice_name=/dev/ttyUSB0
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1661 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1661 ][gsm00 ][-1, 0, 0] controldevice_audio_name=/dev/ttyUSB2
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1663 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1663 ][gsm00 ][-1, 0, 0] gsmopen_serial_sync_period=300
 2012-08-21 09:18:18.973501 [ERR] gsmopen_protocol.cpp:122 rev ccae5cd|07bc7ba[(nil)|37 ][ERRORA 122 ][gsm00 ][-1, 0, 0] port /dev/ttyUSB0, NOT open
 2012-08-21 09:18:18.973501 [ERR] mod_gsmopen.cpp:1670 rev ccae5cd|07bc7ba[(nil)|37 ][ERRORA 1670 ][gsm00 ][-1, 0, 0] gsmopen_serial_init failed
 2012-08-21 09:18:18.973501 [ERR] mod_gsmopen.cpp:1671 rev ccae5cd|07bc7ba[(nil)|37 ][ERRORA 1671 ][gsm00 ][-1, 0, 0] STARTING interface_id=1 FAILED
 2012-08-21 09:18:18.973501 [ERR] mod_gsmopen.cpp:2928 rev ccae5cd|07bc7ba[(nil)|37 ][ERRORA 2928 ][gsm00 ][-1, 0, 0] ALARM on interface gsm00:
 2012-08-21 09:18:18.973501 [CONSOLE] switch_loadable_module.c:1328 Successfully Loaded [mod_gsmopen]
 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'gsmopen'
 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsm'
 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen'
 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen_boost_audio'
 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen_dump'
 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen_sendsms'
 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'sms'

Dongle is voice enabled, PIN request is disabled...What do I do wrong ?

 Cheers / Mike
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From chrisbware at yahoo.it  Tue Aug 21 14:31:55 2012
From: chrisbware at yahoo.it (Chris B. Ware)
Date: Tue, 21 Aug 2012 11:31:55 +0100 (BST)
Subject: [Freeswitch-users] No ringback with Lua Call Control
Message-ID: <1345545115.40493.YahooMailNeo@web132306.mail.ird.yahoo.com>

Hi,

I need a small lua script to originate two extension calls ?and bridge them.
This is my script (based on lua wiki example):

local callee = argv[1]
local called = argv[2]
local domain = argv[3]

Adisp = "None";
param="{ignore_early_media=true,ringback=\'%(1000,4000,425,425)\',origination_caller_id_number="

legA=freeswitch.Session(param..callee.."}user/"..called);
while (Adisp~="ANSWER" and legA:ready()) do
? ? ? ? ?Adisp ?= legA:getVariable("endpoint_disposition");
? ? ? ? os.execute("sleep 1")
end -- waiting for A to answer
if (Adisp=="ANSWER") then
? ? ? ? legB=freeswitch.Session(param..called.."}user/"..callee);
? ? ? ? freeswitch.bridge(legA,legB);
end

The script works but there's no ringback, when A answers and B is ringing.
When B answers, we can talk.

This is my log:
http://pastebin.freeswitch.org/19745



I know that something stupid is missing but I can't work out where's the error!
Any help?
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From ntomer at newgen.co.in  Tue Aug 21 14:58:33 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Tue, 21 Aug 2012 16:28:33 +0530
Subject: [Freeswitch-users] Invalid speech module [cepstral] error on Flite
Message-ID: <007a01cd7f8b$e82d3170$b8879450$@co.in>

Hi,

 

I am trying to make tts work using flite on freeswitch (installed/configured
using this guide - http://wiki.freeswitch.org/wiki/Installation_Guide) on
Ubuntu 12.04.

 

I configured a sample IVR menu using this -
<http://wiki.freeswitch.org/wiki/Examples_ivrmenu_js>
http://wiki.freeswitch.org/wiki/Examples_ivrmenu_js

 

I created a file ivrmenu.js in usr/local/freeswitch/scripts/; and made the
following entry in usr/local/freeswitch/conf/dialplan/default.xml:
 
<extension name="1200">
        <condition field="destination_number" expression="^1200$">
               <action application="javascript" data="ivrmenu.js"/>
        </condition>
</extension>
 

I added the following macro entry in
usr/local/freeswitch/conf/lang/en/en.xml

 

<macro name="mainmenu">
       <input pattern="0">
           <match>
               <action function="speak-text" data="Press 0 for the
Operator."/>
           </match>
       </input>
   
       <input pattern="1">
           <match>
               <action function="speak-text" data="Press 1 for
Accounting."/>
           </match>
       </input>
   
       <input pattern="2">
           <match>
               <action function="speak-text" data="Press 2 for Sales."/>
           </match>
       </input>
   
       <input pattern="3">
           <match>
               <action function="speak-text" data="Press 3 for Support."/>
           </match>
       </input>
   
       <input pattern="4">
           <match>
               <action function="speak-text" data="Press 4 for The
Janitor."/>
           </match>
       </input>
   
       <input pattern="5">
           <match>
               <action function="speak-text" data="Press 5 for The Smoking
Lounge."/>
           </match>
       </input>
   
       <input pattern="6">
           <match>
               <action function="speak-text" data="Press 6 for the men's
bathroom."/>
           </match>
       </input>
   
       <input pattern="7">
           <match>
               <action function="speak-text" data="Press 7 for the women's
powder room."/>
           </match>
       </input>
   
       <input pattern="8">
           <match>
               <action function="speak-text" data="Press 8 for no special
reason."/>
           </match>
       </input>
   
       <input pattern="9">
           <match>
               <action function="speak-text" data="Press 9 for a good
time."/>
           </match>
       </input>
   
       <input pattern="\*">
           <match>
               <action function="speak-text" data="Press star for good
luck."/>
           </match>
       </input>
   
       <input pattern="\#">
           <match>
               <action function="speak-text" data="Press pound for
psychological counseling."/>
           </match>
       </input>
   </macro>
   
   <macro name="saydigits">
       <input pattern="(.*)">
           <match>
               <action function="speak-text" data="You pressed $1"/>
           </match>
       </input>
   </macro>

 

Now when I dial extension 1200, following error is printed on Freeswitch
console -

 

2012-08-21 15:02:16.257525 [NOTICE] switch_channel.c:941 New Channel
sofia/internal/1000 at 192.168.6.109 [187ff238-eb73-11e1-a3ef-ed1990e665f3]

2012-08-21 15:02:16.257525 [INFO] mod_dialplan_xml.c:485 Processing 1000
<1000>->1200 in context default

2012-08-21 15:02:16.297537 [NOTICE] sofia_glue.c:4176 Pre-Answer
sofia/internal/1000 at 192.168.6.109!

2012-08-21 15:02:16.317513 [NOTICE] mod_spidermonkey.c:2098 Channel
[sofia/internal/1000 at 192.168.6.109] has been answered

2012-08-21 15:02:16.341524 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]!

2012-08-21 15:02:16.341524 [ERR] switch_ivr_play_say.c:2466 Invalid TTS
module!

2012-08-21 15:02:21.625528 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]!

2012-08-21 15:02:21.625528 [ERR] switch_ivr_play_say.c:2466 Invalid TTS
module!

2012-08-21 15:02:21.737528 [NOTICE] switch_core_state_machine.c:249
sofia/internal/1000 at 192.168.6.109 has executed the last dialplan
instruction, hanging up.

2012-08-21 15:02:21.737528 [NOTICE] switch_core_state_machine.c:251 Hangup
sofia/internal/1000 at 192.168.6.109 [CS_EXECUTE] [NORMAL_CLEARING]

2012-08-21 15:02:21.917533 [NOTICE] switch_core_session.c:1447 Session 1
(sofia/internal/1000 at 192.168.6.109) Ended

2012-08-21 15:02:21.917533 [NOTICE] switch_core_session.c:1449 Close Channel
sofia/internal/1000 at 192.168.6.109 [CS_DESTROY]

2012-08-21 15:02:32.957527 [NOTICE] switch_channel.c:941 New Channel
sofia/internal/1000 at 192.168.6.109 [2274b6a2-eb73-11e1-a3f5-ed1990e665f3]

2012-08-21 15:02:32.957527 [INFO] mod_dialplan_xml.c:485 Processing 1000
<1000>->1200 in context default

2012-08-21 15:02:32.978655 [NOTICE] sofia_glue.c:4176 Pre-Answer
sofia/internal/1000 at 192.168.6.109!

2012-08-21 15:02:32.978655 [NOTICE] mod_spidermonkey.c:2098 Channel
[sofia/internal/1000 at 192.168.6.109] has been answered

2012-08-21 15:02:33.005538 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]!

2012-08-21 15:02:33.005538 [ERR] switch_ivr_play_say.c:2466 Invalid TTS
module!

2012-08-21 15:02:39.124530 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]!

2012-08-21 15:02:39.124530 [ERR] switch_ivr_play_say.c:2466 Invalid TTS
module!

2012-08-21 15:02:45.265517 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]!

2012-08-21 15:02:45.265517 [ERR] switch_ivr_play_say.c:2466 Invalid TTS
module!

2012-08-21 15:02:51.385575 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]!

2012-08-21 15:02:51.385575 [ERR] switch_ivr_play_say.c:2466 Invalid TTS
module!

2012-08-21 15:02:51.497528 [NOTICE] switch_core_state_machine.c:249
sofia/internal/1000 at 192.168.6.109 has executed the last dialplan
instruction, hanging up.

2012-08-21 15:02:51.497528 [NOTICE] switch_core_state_machine.c:251 Hangup
sofia/internal/1000 at 192.168.6.109 [CS_EXECUTE] [NORMAL_CLEARING]

2012-08-21 15:02:51.497528 [NOTICE] switch_core_session.c:1447 Session 2
(sofia/internal/1000 at 192.168.6.109) Ended

2012-08-21 15:02:51.497528 [NOTICE] switch_core_session.c:1449 Close Channel
sofia/internal/1000 at 192.168.6.109 [CS_DESTROY]

 

What seems to be the problem here? Am I missing something? Please help.

 

Thanks

 

Nitin

 


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From mi.ke at null.net  Tue Aug 21 15:18:08 2012
From: mi.ke at null.net (Mi Ke)
Date: Tue, 21 Aug 2012 07:18:08 -0400
Subject: [Freeswitch-users] mod_gsmopen serial init failed
Message-ID: <20120821111808.210950@gmx.com>

It appears that selinux policies prevented freeswitch user from accessing /dev/ttyUSB*

 [root at freeswitch ~]# ls -l /dev/ttyUSB*
 crw-rw----. 1 root dialout 188, 0 Aug 21 10:57 /dev/ttyUSB0
 crw-rw----. 1 root dialout 188, 1 Aug 21 10:48 /dev/ttyUSB1
 crw-rw----. 1 root dialout 188, 2 Aug 21 10:48 /dev/ttyUSB2

 so adding freeswitch user to dialout group by running "usermod -a -G dialout freeswitch" worked for me.

 Have a nice day
 Mike
----- Original Message -----
From: Mi Ke
Sent: 08/21/12 12:58 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] mod_gsmopen serial init failed

Hi All,

 I have problems getting mod_gsmopen to see my E1550. Both minicom and wvdial see the dongle on /dev/ttyUSB0:

 ttyUSB0<*1>: ATQ0 V1 E1 -- OK
 ttyUSB0<*1>: ATQ0 V1 E1 Z -- OK
 ttyUSB0<*1>: ATQ0 V1 E1 S0=0 -- OK
 ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 -- OK
 ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 -- OK
 ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0 -- OK
 ttyUSB0<*1>: Modem Identifier: ATI -- Manufacturer: huawei
 ttyUSB0<*1>: Speed 9600: AT -- OK
 ttyUSB0<*1>: Max speed is 9600; that should be safe.
 ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0 -- OK

 Welcome to minicom 2.3

 OPTIONS: I18n
 Compiled on Aug 19 2010, 05:50:19.
 Port /dev/ttyUSB0

 Press CTRL-A Z for help on special keys
 AT S7=45 S0=0 L1 V1 X4 &c1 E1 Q0
 OK 

 at+cpin?
 +CPIN: READY
 OK

 but mod_gsmopen does not want to see it:

 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1203 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1203 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 0 ?????
 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1204 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1204 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 1 ???^?
 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1205 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1205 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 2 ???????
 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1206 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1206 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 3 ??????
 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1207 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1207 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 4 ???
 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1208 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1208 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 5 ??
 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1209 rev ccae5cd|07bc7ba[(nil)|37 ][NOTICA 1209 ][none ][-1,-1,-1] GSMOPEN Charset Output Test 6 ??
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1226 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1226 ][none ][-1,-1,-1] globals.debug=0
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1228 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1228 ][none ][-1,-1,-1] globals.debug=8
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1234 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1234 ][none ][-1,-1,-1] globals.dialplan=XML
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1240 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1240 ][none ][-1,-1,-1] globals.context=external
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1231 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1231 ][none ][-1,-1,-1] globals.hold_music=
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1237 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1237 ][none ][-1,-1,-1] globals.destination=5000
 2012-08-21 09:18:18.973501 [WARNING] mod_gsmopen.cpp:1653 rev ccae5cd|07bc7ba[(nil)|37 ][WARNINGA 1653 ][gsm00 ][-1, 0, 0] STARTING interface_id=1
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1654 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1654 ][gsm00 ][-1, 0, 0] id=1
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1655 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1655 ][gsm00 ][-1, 0, 0] name=gsm00
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1656 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1656 ][gsm00 ][-1, 0, 0] hold-music=
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1657 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1657 ][gsm00 ][-1, 0, 0] context=default
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1658 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1658 ][gsm00 ][-1, 0, 0] dialplan=XML
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1659 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1659 ][gsm00 ][-1, 0, 0] destination=5000
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1660 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1660 ][gsm00 ][-1, 0, 0] controldevice_name=/dev/ttyUSB0
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1661 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1661 ][gsm00 ][-1, 0, 0] controldevice_audio_name=/dev/ttyUSB2
 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1663 rev ccae5cd|07bc7ba[(nil)|37 ][DEBUG_GSMOPEN 1663 ][gsm00 ][-1, 0, 0] gsmopen_serial_sync_period=300
 2012-08-21 09:18:18.973501 [ERR] gsmopen_protocol.cpp:122 rev ccae5cd|07bc7ba[(nil)|37 ][ERRORA 122 ][gsm00 ][-1, 0, 0] port /dev/ttyUSB0, NOT open
 2012-08-21 09:18:18.973501 [ERR] mod_gsmopen.cpp:1670 rev ccae5cd|07bc7ba[(nil)|37 ][ERRORA 1670 ][gsm00 ][-1, 0, 0] gsmopen_serial_init failed
 2012-08-21 09:18:18.973501 [ERR] mod_gsmopen.cpp:1671 rev ccae5cd|07bc7ba[(nil)|37 ][ERRORA 1671 ][gsm00 ][-1, 0, 0] STARTING interface_id=1 FAILED
 2012-08-21 09:18:18.973501 [ERR] mod_gsmopen.cpp:2928 rev ccae5cd|07bc7ba[(nil)|37 ][ERRORA 2928 ][gsm00 ][-1, 0, 0] ALARM on interface gsm00:
 2012-08-21 09:18:18.973501 [CONSOLE] switch_loadable_module.c:1328 Successfully Loaded [mod_gsmopen]
 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:146 Adding Endpoint 'gsmopen'
 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsm'
 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen'
 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen_boost_audio'
 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen_dump'
 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding API Function 'gsmopen_sendsms'
 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:403 Adding Chat interface 'sms'

Dongle is voice enabled, PIN request is disabled...What do I do wrong ?

 Cheers / Mike
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From jaasmailing at gmail.com  Tue Aug 21 16:59:26 2012
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Tue, 21 Aug 2012 14:59:26 +0200
Subject: [Freeswitch-users] Realm and gateway authentication
Message-ID: <5033862E.3090105@gmail.com>

Hi all,

I have a trunk with a gateway configured with name/proxy "sip.xxx.com".
On inbound calls the gateway sends sip packets with a realm "1.2.3.4.5" 
which is the resolution of sip.xxx.com.

Unfortunately when the gateway asks for the BYE authentication, 
freeswitch reply with:

2012-08-21 14:51:47.291432 [ERR] sofia_reg.c:2167 Cannot locate any 
authentication credentials to complete an authentication request for 
realm '"1.2.3.4.5"'

Is there any way to tell freeswitch to use the reverse DNS resolution 
for the matching of realm?
If I set directly the IP, all is fine.


Best regards,
Carlo Dimaggio
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From ziga.jakhel at integrum.si  Tue Aug 21 13:19:34 2012
From: ziga.jakhel at integrum.si (=?iso-8859-2?Q?=AEiga_Jakhel?=)
Date: Tue, 21 Aug 2012 09:19:34 +0000
Subject: [Freeswitch-users] ERROR! Unable to open ZRTP cache file
Message-ID: <E0A83136CCF23E44B603A73AB66E76390D03EA44@HAIL.hq.integrum-it.com>

Hi, everyone!

In my freeswitch installation (Windows x64 build, running on Win 2008 R2 x64 Std), I get the following sequence roughly every 15 minutes:

2012-08-21 11:13:42.018826 [DEBUG] switch_rtp.c:915  [  zrtp cache]: Storing ZRTP Cache...
2012-08-21 11:13:42.018826 [DEBUG] switch_rtp.c:915  [  zrtp cache]:   Storing ZRTP cache to <>...
2012-08-21 11:13:42.018826 [DEBUG] switch_rtp.c:915  [  zrtp cache]:   ERROR! unable to open ZRTP cache file <>.
2012-08-21 11:13:42.018826 [DEBUG] switch_rtp.c:915  [  zrtp cache]: Storing ZRTP Cache - DONE.
2012-08-21 11:13:42.018826 [DEBUG] switch_rtp.c:776 Saving ZRTP cache: OK

Any ideas what this is about and if it is something I should do something about?

Regards,


?iga Jakhel, MSc

Integrum d.o.o.
Smoletova 6, 1230 Dom?ale
Slovenia, EU
t: +386 820 50505
f: +386 820 50504
m: +386 41 690568
e: ziga.jakhel at integrum.si<mailto:ziga.jakhel at integrum.si>

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From roger.castaldo at gmail.com  Tue Aug 21 18:23:31 2012
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Tue, 21 Aug 2012 10:23:31 -0400
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Project Looking
 For Volunteers: Web Site Dev/Maint
In-Reply-To: <CAMbW9o+94bnu6ntg4nv+jYVgxOGvk8UmisMm6e0SBVZTKNYmuA@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
	<CAMwZ-t+6xdX6wwwE7vUmvaHS9D50FN5o5iR4Qz3y+MvYzNr-Pw@mail.gmail.com>
	<CAF+3j-tMcwmYkUQQ1RV=Mo72d6BU2gYDyU7BT5P3eKe_9OYT8A@mail.gmail.com>
	<CAMbW9o+94bnu6ntg4nv+jYVgxOGvk8UmisMm6e0SBVZTKNYmuA@mail.gmail.com>
Message-ID: <CAAYf2LtWA4qpg1zwdJzz+WOQ=iyQ7iQ8tWfQoqPSHgZc0ZkT4A@mail.gmail.com>

Guess I should have noted that i have been playing around a fair bit with
jquery mobile and redesigned one of my sites to be mobile compliant as
well, in case some mobile browser stuff is needed

On Tue, Aug 21, 2012 at 5:13 AM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> Same as Gabe, except my design abilities are shockingly bad.
>
>
> On 21 August 2012 08:56, Antonio Teixeira <eagle.antonio at gmail.com> wrote:
>
>> Hello guys
>> Regarding the Django / Python / Apache .
>>
>> Count me in :)
>>
>> Regards
>> Ant?nio Teixeira
>>
>>
>> 2012/8/21 Gabriel Gunderson <gabe at gundy.org>
>>
>>> On Mon, Aug 20, 2012 at 3:45 PM, Michael Collins <msc at freeswitch.org>
>>> wrote:
>>> > HTML5
>>> > CSS2/3
>>> > Javascript/JQuery
>>> > Django/Python
>>> > Web design/graphics
>>> > Apache administration in LAMP environment
>>>
>>> The folks at Izeni are happy to help where we can (with the trimmed
>>> list of techs).  We feel it's one of the ways we can give back to the
>>> FreeSWITCH project and community that we benefit so much from.
>>>
>>> We're look forward to working with the others who join in.
>>>
>>>
>>> Best,
>>> Gabe
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Aug 21 19:01:37 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 21 Aug 2012 08:01:37 -0700
Subject: [Freeswitch-users] Initiating application jitterbuffer multiple
	times
In-Reply-To: <CAKbxfG-Mo+7wdtYzpODH6K3Sh=V-SnyqRWurkw72zTjAvpQK+g@mail.gmail.com>
References: <CABaNFCZzw_jdEMUo3v6JfVbKO=9kKGdNLsKRmkEM8QZ963AL2g@mail.gmail.com>
	<CAD-qRkhuEBUuD+JBpJ4K1osbdEfvoOUCbg56vQYNgWf9SXyKxQ@mail.gmail.com>
	<CABaNFCZ5ymSALKMPztxZYxFWQM03thh-6MJubpfnQ6HfCQnhqw@mail.gmail.com>
	<CAKbxfG-Mo+7wdtYzpODH6K3Sh=V-SnyqRWurkw72zTjAvpQK+g@mail.gmail.com>
Message-ID: <CAKzWOxW7hXCQxzJAnVeYFK1n=B6ENERzvrajeQWKGmH8EkgNdQ@mail.gmail.com>

FYI, this is very useful information so I tossed it up on the wiki on the
jitterbuffer page:
http://wiki.freeswitch.org/wiki/Jitterbuffer#Interesting_Information

If anyone feels like making it look prettier by all means do so. :)

-Michael

On Mon, Aug 20, 2012 at 10:25 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> If the jitterbuffer is already on, calling it again will just resize
> it, so setting it to the same is redundant but harmless.
>
> If you are surprised by why the jitterbuffer is paused during bridge:
>
> If both sides of a bridge are RTP and both sides have a jb, its fairly
> useless.  In fact if anything, it can worsen call quality.
>
> You should only run jitterbuffers at points of termination change of
> protocol.  Examples, if FS was hosting a conference or IVR, if you are
> bridging the call to a phone for instance, you want to not use a
> jitterbuffer because you want to preserve the original timestamps so
> your phone can use its own jitterbuffer.
>
> For special examples where you are using FS jitterbuffer in front of
> something else that may not have one or some other special
> circumstance you can use the setting chris mentioned to leave it
> running.
>
>
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From x.liu at hw.ac.uk  Tue Aug 21 21:30:25 2012
From: x.liu at hw.ac.uk (x.liu)
Date: Tue, 21 Aug 2012 18:30:25 +0100
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <2526639.vT3cT5qGKG@virtex>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<503226CE.7040307@hw.ac.uk> <50323D6A.3020708@anew.com.ve>
	<2526639.vT3cT5qGKG@virtex>
Message-ID: <5033C5B1.3070509@hw.ac.uk>

Hi g & Dario,

Many thanks for your suggestions and explanations! Very helpful!

I should have made my scenarios clearer in the first place:
My agents are IVR apps which have registered to FS and are always 
available unless they are answering calls.


Cheers,
Xing


On 08/20/2012 05:32 PM, g wrote:
> Thanks Dario. You gave the answer I would like to do.
> Liu, You can avoid agents dialing extensions to login/logout/pause, but at
> least they have to click on some web app to let the system knows they are
> available.
> Another option is to let the system considerthem always available even when
> the softphones are'nt registered. If you set a correct distribution criteria,
> the unregistered softphones are skipped in a round robin distribution, as the
> agents were not logged in. But in this case, forget about statistics on drops,
> presence, waiting time on queue etc.
> g
>
> On Monday 20 August 2012 09:06:42 Saugort Dario Garcia Tovar wrote:
>> mmm,
>>
>> You think that the agent is dialing an extension. It is not what is
>> happening. The agent is dialing a number that invoke a mod_callcenter
>> function: login/logout/not ready. As FS is like a pbx (and more than a
>> pbx), it seems like an extension.
>>
>> Just, registrer to FS is not enough. What happen when the agent need to
>> go the restroom, take coffee, couching, or take a break? Then, you will
>> need to reproduce in the agent gui the funcion of login, logout and
>> break... but again, what happen if your agent gui fail? your callcenter
>> service will stop because your gui fail? Also, if you just use register
>> signaling, you could face a wide sort of operational issues like, what
>> happen if the agent left the position and left the their seats: then
>> calls will be routed to empty seats!
>>
>> If you want to get the agent login/logout without the agent dial
>> nothing... if I am not wrong, mod_callcenter offer functions to log and
>> logout agent from fs console, so I think you can invoke them using
>> execute() from lua or your script.
>>
>> On 8/20/2012 7:30 AM, x.liu wrote:
>>> Hi g,
>>>
>>> Thanks for your response!
>>>
>>> My situation is that I don't want the agent to dial the extension to be
>>> able to accept calls.
>>> The agents (e.g. a FS softphone client, or a Java application) just need
>>> to register to FS as SIP clients.
>>>
>>> And I've not figured out what commands I should send if I use extension
>>> "^agent-login$" and FS socket.
>>>
>>> So I guess I may be better not to use mod_callcenter for routing calls
>>> to different extensions.
>>> just to use FS ESL and my own logics to bridge the calls.
>>>
>>> I was curious about the reply "You don't need agent to dial anything to
>>> get the call from queue... "
>>> and I thought my question is relevant to the thread, so I asked
>>> questions in this thread.
>>> Apologies for that. I could have opened a different thread for the
>>> questions :-)
>>>
>>> Cheers,
>>> Xing
>>>
>>> On 08/20/2012 11:53 AM, g wrote:
>>>> You can log an agent into a queue by a web application in which a
>>>> button click handles commands via the FS socket, or just let the
>>>> agent dial an extension (i.e. 999+itssoftphonenumber) where dialplan
>>>> sets that agent in the correct state.
>>>> ie (replace agent_login regex into something like 999201 if your agent
>>>> is 201)<extension name="agent_login">
>>>>
>>>>      <condition field="destination_number"
>>>>      expression="^agent-login$">
>>>>
>>>>        <action application="set"
>>>>        data="res=${callcenter_config(agent set status>>
>>>> ${caller_id_number}@${domain_name} 'Available')}" />
>>>>
>>>>        <action application="answer" data=""/>
>>>>        <action application="sleep" data="500"/>
>>>>        <action application="playback"
>>>>        data="ivr/ivr-you_are_now_logged_in.wav"/>  <action
>>>>        application="hangup" data=""/>
>>>>
>>>>      </condition>
>>>>
>>>> </extension>
>>>>
>>>> <extension name="agent_logoff">
>>>>
>>>>      <condition field="destination_number"
>>>>      expression="^agent-logoff$">
>>>>
>>>>        <action application="set"
>>>>        data="res=${callcenter_config(agent set status>>
>>>> ${caller_id_number}@${domain_name} 'Logged Out')}" />
>>>>
>>>>        <action application="answer" data=""/>
>>>>        <action application="sleep" data="500"/>
>>>>        <action application="playback"
>>>>        data="ivr/ivr-you_are_now_logged_out.wav"/>  <action
>>>>        application="hangup" data=""/>
>>>>
>>>>      </condition>
>>>>
>>>> </extension>
>>>>
>>>> The important thing is tha the agent is, somehow, putted in
>>>> "available"
>>>> status. You can also do it via fs_cli, but now I can't remeber the
>>>> exact
>>>> syntax for that. Help from cli should clarify.
>>>>
>>>> At this point, the agent should result in "waiting" state
>>>> Of course, you should before declared that agent as member of a tier
>>>> or, at least, of a queue.
>>>>
>>>> Hope it helps
>>>> g
>>>>
>>>> On Friday 17 August 2012 14:48:04 x.liu wrote:
>>>>> I am also trying the mod_callcenter. What I want is to route a call
>>>>> to
>>>>> one agent ( a registered FS client, e.g. a SIP softphone)
>>>>> of a list of agents by a Random or RoundRobin policy.
>>>>>
>>>>> When I dial the extension which calls the callcenter app, I only
>>>>> have
>>>>> hold-on music. It does not route a call to a agent.
>>>>> You said " You don't need agent to dial anything to get the call
>>>>> from
>>>>> queue. ",
>>>>> I am wondering how I could let an agent log into the queue?
>>>>>
>>>>> Thanks!
>>>>>
>>>>> Xing
>>>>>
>>>>> On 08/17/2012 01:56 PM, g wrote:
>>>>>> Me too, I'm often dealing with callcenters (inbound+outbound). I
>>>>>> found
>>>>>> in
>>>>>> Freeswitch the easyest solution to manage queue and agents. I'm
>>>>>> not a
>>>>>> programmer, but I have experience with Asterisk, Freeswitch and
>>>>>> the
>>>>>> whole
>>>>>> workflow of complex callcenters.
>>>>>> My suggestion too is to use mod_callcenter. Easy, fast, powerful.
>>>>>> You
>>>>>> don't need agent to dial anything to get the call from queue.
>>>>>> In my opinion, you can use that module or, for more complex things
>>>>>> (ie. outbound predictive) you can use the socket manager to
>>>>>> perform  call placement, call transfer, conference with
>>>>>> thirdy-part recording  IVR etc.
>>>>>> I feel the best solution, also if it will require some effort for
>>>>>> a Java programmer, is to do all that in php, which is light and
>>>>>> fast also for older PC, able to run also with thin-client low
>>>>>> memory low-cpu machines. If you can plan to turn to php and you
>>>>>> don't have too short time, I can consider to help for the
>>>>>> Freeswitch part.
>>>>>> giuliano
>>>>>>
>>>>>> On Thursday 16 August 2012 17:10:33 Nitin Tomer wrote:
>>>>>>> Hi,
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> I am working on a Contact Center solution. It will support mail,
>>>>>>> chat
>>>>>>> and call queries.
>>>>>>>
>>>>>>> The requirements are:
>>>>>>>
>>>>>>> 1. An end-customer calls, the call is handled by FreeSWITCH Auto
>>>>>>> Attendant. 2. Customer is presented with a menu and makes
>>>>>>> selection.
>>>>>>> His call is put on hold and an entry is made in my system's
>>>>>>> database
>>>>>>> for incoming queries. 3. These queries are shown to agents
>>>>>>> handling
>>>>>>> calls.
>>>>>>> 4. And agent clicks on a query, he is shown an extension where
>>>>>>> call is parked. He dials that and is connected to the customer.
>>>>>>>
>>>>>>> 5. He talks to the customer and resolve his queries.
>>>>>>>
>>>>>>> Please guide me on how can do it. The application will be
>>>>>>> written in
>>>>>>> Java.
>>>>>>>
>>>>>>> I am an experienced programmers in Java/J2EE; but doesn't have
>>>>>>> much
>>>>>>> knowledge of VoIP/FreeSWITCH. I've configured FreeSWITCH on
>>>>>>> Ubuntu
>>>>>>> 12.04; and have installed X-Lite Softphones on two Windows
>>>>>>> machines.
>>>>>>> I configured these phones to work with FreeSWITCH and they are
>>>>>>> working fine.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> I think that I'd have to use Valet Parking to park customer's
>>>>>>> call on
>>>>>>> an
>>>>>>> extension, then pass that extension to agent who will dial it
>>>>>>> and will be connected to the customer.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Please tell me whether this approach will work? And how should I
>>>>>>> go
>>>>>>> about it.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> All help would be much appreciated.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Regards
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Nitin
>>>>>> __________________________________________________________________
>>>>>> ______ _
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>> ______________________________________________________________________
>>>> ___ Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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-use
>>>> rs
>>>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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



-- 
Heriot-Watt University is the Sunday Times
Scottish University of the Year 2011-2012

We invite research leaders and ambitious early career researchers to 
join us in leading and driving research in key inter-disciplinary themes. 
Please see www.hw.ac.uk/researchleaders for further information and how
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Heriot-Watt University is a Scottish charity
registered under charity number SC000278.



From miconda at gmail.com  Tue Aug 21 21:57:26 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 21 Aug 2012 19:57:26 +0200
Subject: [Freeswitch-users] sip profile - accept-blind-auth vs
Message-ID: <5033CC06.2040605@gmail.com>

Hello,

in the past I used to set:

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

in the sip profile in order to skip user authentication for calls.

Lately I started to play a bit with 1.2 stable branch and seems that 
setting auth-calls to false is no longer doing what I expected, calls 
being challenged for user authentication.

Setting instead the accept-blind-auth to false got me what I wanted, like:

     <!-- accept any authentication without actually checking (not a 
good feature for most people) -->
     <param name="accept-blind-auth" value="true"/>

But from the comment (checked the wiki as well, but has the same text) 
is a bit unclear what is the real purpose for it.

Isn't auth-calls=false supposed to accept calls without user 
authentication anymore?

For this particular case, I play some announcements, like 'user not 
available', and should work also for calls coming from outside. The 
access is restricted by IP address ACL, allowing SIP traffic only from 
my Kamailio instance.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From gmaruzz at gmail.com  Tue Aug 21 22:07:09 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 21 Aug 2012 20:07:09 +0200
Subject: [Freeswitch-users] mod_gsmopen: ROAMING Network ERROR
In-Reply-To: <CAMjaK3B-Y+nswvVnj49Jh_+sZCJKvcXiSmhJ7ksMWo6UXbGt9Q@mail.gmail.com>
References: <CAD=Wdqi_rbz3Z0XrYcS_CP29wSV94G8R7cJpHKRqZoeNorhmxw@mail.gmail.com>
	<CAMjaK3B-Y+nswvVnj49Jh_+sZCJKvcXiSmhJ7ksMWo6UXbGt9Q@mail.gmail.com>
Message-ID: <CALXCt0q+CZsOBL6Uv0_W9TMje+qZGb+rguej-cgMqRNQA9mo6w@mail.gmail.com>

hi Sameer,
I'm in holidays.

If problems persists, please contact me after Sept 10th.

Dimitri, you're right. Btw, you seems very knowledgeable about gsmopen
and related stuff. Please feel free to modify expand mod-gsmopen wiki
page and/or send patches to jira. Thanks in advance Dimitri!

-giovanni

On 8/21/12, Dmitry Lysenko <dvl36.ripe.nick at gmail.com> wrote:
> Hello.
> I have the same "ROAMING Network" ALERTs, it's annoying, but everything
> working OK.
> Checked with  Huawei E1550.
>
> P.S. Are you sure that your Huawei E169  is voice enabled?
>
> 2012/8/21 Josep Maria Vi?olas <vinolasbcn at gmail.com>
>
>> I have a Huawei E169 with a SIM from a "virtual provider" that uses other
>> provider's networks, so it is allways technically 'roaming'.
>>
>> With mod_gsmopen the problem is that throws an error about "ROAMING
>> Network" and also an ALERT. I don't know if that is why it is not able to
>> make calls. In that case, how can I tell the module that roaming is not a
>> bad thing??
>>
>> This is freeswitch debug output when I load mod_gsmmobile:
>> ------------------------------------------
>>
> ...
>
>> ------------------------------------------
>> ** From extension 1000 I hear a calling tone even the Freeswitch log says
>> that ATD+N? command failed.
>>
>> Anyone knows if the problem is the ROAMING Network error??
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 Hector.Geraldino at ipsoft.com  Tue Aug 21 22:07:24 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Tue, 21 Aug 2012 14:07:24 -0400
Subject: [Freeswitch-users] Invalid speech module [cepstral] error on
 Flite
In-Reply-To: <007a01cd7f8b$e82d3170$b8879450$@co.in>
References: <007a01cd7f8b$e82d3170$b8879450$@co.in>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD073F6F2858@NY1-EXMB-01.ip-soft.net>

Hi,

Do you have cepstral binaries?
Have you uncommented the mod_cepstral  entry on the modules.conf.xml file in conf/autoload_config?
If your intention is to use flite, have you changed the value for the tts-engine in the en.xml file (conf/lang/en/en.xml) ?


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin Tomer
Sent: Tuesday, August 21, 2012 6:59 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Invalid speech module [cepstral] error on Flite

Hi,

I am trying to make tts work using flite on freeswitch (installed/configured using this guide - http://wiki.freeswitch.org/wiki/Installation_Guide) on Ubuntu 12.04.

I configured a sample IVR menu using this - http://wiki.freeswitch.org/wiki/Examples_ivrmenu_js


I created a file ivrmenu.js in usr/local/freeswitch/scripts/; and made the following entry in usr/local/freeswitch/conf/dialplan/default.xml:



<extension name="1200">

        <condition field="destination_number" expression="^1200$">

               <action application="javascript" data="ivrmenu.js"/>

        </condition>

</extension>


I added the following macro entry in usr/local/freeswitch/conf/lang/en/en.xml


<macro name="mainmenu">

       <input pattern="0">

           <match>

               <action function="speak-text" data="Press 0 for the Operator."/>

           </match>

       </input>



       <input pattern="1">

           <match>

               <action function="speak-text" data="Press 1 for Accounting."/>

           </match>

       </input>



       <input pattern="2">

           <match>

               <action function="speak-text" data="Press 2 for Sales."/>

           </match>

       </input>



       <input pattern="3">

           <match>

               <action function="speak-text" data="Press 3 for Support."/>

           </match>

       </input>



       <input pattern="4">

           <match>

               <action function="speak-text" data="Press 4 for The Janitor."/>

           </match>

       </input>



       <input pattern="5">

           <match>

               <action function="speak-text" data="Press 5 for The Smoking Lounge."/>

           </match>

       </input>



       <input pattern="6">

           <match>

               <action function="speak-text" data="Press 6 for the men's bathroom."/>

           </match>

       </input>



       <input pattern="7">

           <match>

               <action function="speak-text" data="Press 7 for the women's powder room."/>

           </match>

       </input>



       <input pattern="8">

           <match>

               <action function="speak-text" data="Press 8 for no special reason."/>

           </match>

       </input>



       <input pattern="9">

           <match>

               <action function="speak-text" data="Press 9 for a good time."/>

           </match>

       </input>



       <input pattern="\*">

           <match>

               <action function="speak-text" data="Press star for good luck."/>

           </match>

       </input>



       <input pattern="\#">

           <match>

               <action function="speak-text" data="Press pound for psychological counseling."/>

           </match>

       </input>

   </macro>



   <macro name="saydigits">

       <input pattern="(.*)">

           <match>

               <action function="speak-text" data="You pressed $1"/>

           </match>

       </input>

   </macro>

Now when I dial extension 1200, following error is printed on Freeswitch console -

2012-08-21 15:02:16.257525 [NOTICE] switch_channel.c:941 New Channel sofia/internal/1000 at 192.168.6.109<mailto:sofia/internal/1000 at 192.168.6.109> [187ff238-eb73-11e1-a3ef-ed1990e665f3]
2012-08-21 15:02:16.257525 [INFO] mod_dialplan_xml.c:485 Processing 1000 <1000>->1200 in context default
2012-08-21 15:02:16.297537 [NOTICE] sofia_glue.c:4176 Pre-Answer sofia/internal/1000 at 192.168.6.109<mailto:sofia/internal/1000 at 192.168.6.109>!
2012-08-21 15:02:16.317513 [NOTICE] mod_spidermonkey.c:2098 Channel [sofia/internal/1000 at 192.168.6.109] has been answered
2012-08-21 15:02:16.341524 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]!
2012-08-21 15:02:16.341524 [ERR] switch_ivr_play_say.c:2466 Invalid TTS module!
2012-08-21 15:02:21.625528 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]!
2012-08-21 15:02:21.625528 [ERR] switch_ivr_play_say.c:2466 Invalid TTS module!
2012-08-21 15:02:21.737528 [NOTICE] switch_core_state_machine.c:249 sofia/internal/1000 at 192.168.6.109<mailto:sofia/internal/1000 at 192.168.6.109> has executed the last dialplan instruction, hanging up.
2012-08-21 15:02:21.737528 [NOTICE] switch_core_state_machine.c:251 Hangup sofia/internal/1000 at 192.168.6.109<mailto:sofia/internal/1000 at 192.168.6.109> [CS_EXECUTE] [NORMAL_CLEARING]
2012-08-21 15:02:21.917533 [NOTICE] switch_core_session.c:1447 Session 1 (sofia/internal/1000 at 192.168.6.109<mailto:sofia/internal/1000 at 192.168.6.109>) Ended
2012-08-21 15:02:21.917533 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/1000 at 192.168.6.109<mailto:sofia/internal/1000 at 192.168.6.109> [CS_DESTROY]
2012-08-21 15:02:32.957527 [NOTICE] switch_channel.c:941 New Channel sofia/internal/1000 at 192.168.6.109<mailto:sofia/internal/1000 at 192.168.6.109> [2274b6a2-eb73-11e1-a3f5-ed1990e665f3]
2012-08-21 15:02:32.957527 [INFO] mod_dialplan_xml.c:485 Processing 1000 <1000>->1200 in context default
2012-08-21 15:02:32.978655 [NOTICE] sofia_glue.c:4176 Pre-Answer sofia/internal/1000 at 192.168.6.109<mailto:sofia/internal/1000 at 192.168.6.109>!
2012-08-21 15:02:32.978655 [NOTICE] mod_spidermonkey.c:2098 Channel [sofia/internal/1000 at 192.168.6.109] has been answered
2012-08-21 15:02:33.005538 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]!
2012-08-21 15:02:33.005538 [ERR] switch_ivr_play_say.c:2466 Invalid TTS module!
2012-08-21 15:02:39.124530 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]!
2012-08-21 15:02:39.124530 [ERR] switch_ivr_play_say.c:2466 Invalid TTS module!
2012-08-21 15:02:45.265517 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]!
2012-08-21 15:02:45.265517 [ERR] switch_ivr_play_say.c:2466 Invalid TTS module!
2012-08-21 15:02:51.385575 [ERR] switch_core_speech.c:61 Invalid speech module [cepstral]!
2012-08-21 15:02:51.385575 [ERR] switch_ivr_play_say.c:2466 Invalid TTS module!
2012-08-21 15:02:51.497528 [NOTICE] switch_core_state_machine.c:249 sofia/internal/1000 at 192.168.6.109<mailto:sofia/internal/1000 at 192.168.6.109> has executed the last dialplan instruction, hanging up.
2012-08-21 15:02:51.497528 [NOTICE] switch_core_state_machine.c:251 Hangup sofia/internal/1000 at 192.168.6.109<mailto:sofia/internal/1000 at 192.168.6.109> [CS_EXECUTE] [NORMAL_CLEARING]
2012-08-21 15:02:51.497528 [NOTICE] switch_core_session.c:1447 Session 2 (sofia/internal/1000 at 192.168.6.109<mailto:sofia/internal/1000 at 192.168.6.109>) Ended
2012-08-21 15:02:51.497528 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/1000 at 192.168.6.109<mailto:sofia/internal/1000 at 192.168.6.109> [CS_DESTROY]

What seems to be the problem here? Am I missing something? Please help.

Thanks

Nitin


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From g.d.monnezza at tiscali.it  Tue Aug 21 22:14:42 2012
From: g.d.monnezza at tiscali.it (g)
Date: Tue, 21 Aug 2012 20:14:42 +0200
Subject: [Freeswitch-users] Help with settings
In-Reply-To: <5032C40F.9050302@pripojtese.net>
References: <5032C40F.9050302@pripojtese.net>
Message-ID: <1575699.evKPYiHUWV@virtex>

I think it would be better to specify what this "few things" deal about :) ... 
Dialplan? NAT configuration? Voicemail? Or ... 

g

On Tuesday 21 August 2012 00:11:11 Jakub Tencl wrote:
> Hello,
> 
> is here anyone who is willing to help me with a few things on my
> freeswitch server remotely?
> 
> Thanks
> 
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 vbvbrj at gmail.com  Tue Aug 21 23:02:32 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Tue, 21 Aug 2012 22:02:32 +0300
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <7518514.pVGd9TpNrB@virtex>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<10440306.PhsNeVk7eG@virtex> <502E4B94.1040503@hw.ac.uk>
	<7518514.pVGd9TpNrB@virtex>
Message-ID: <5033DB48.5090006@gmail.com>

For my callcenter I use keypad's numbers for agent's login/logout. For 
example, my callcenter has extension 1011, then I use:

  <extension name="agent_login">
     <condition field="destination_number" expression="^agent-login|*1011$">
       <action application="set" data="res=${callcenter_config(agent set 
status
  ${caller_id_number}@${domain_name} 'Available')}" />
       <action application="answer" data=""/>
       <action application="sleep" data="500"/>
       <action application="playback" 
data="ivr/ivr-you_are_now_logged_in.wav"/>
       <action application="hangup" data=""/>
     </condition>
  </extension>

  <extension name="agent_logoff">
     <condition field="destination_number" 
expression="^agent-logoff|#1011$">
       <action application="set" data="res=${callcenter_config(agent set 
status
  ${caller_id_number}@${domain_name} 'Logged Out')}" />
       <action application="answer" data=""/>
       <action application="sleep" data="500"/>
       <action application="playback" 
data="ivr/ivr-you_are_now_logged_out.wav"/>
       <action application="hangup" data=""/>
     </condition>
  </extension>

Ofcourse, phones must be setup so *1011 and #1011 are send to FS and not 
intepreted by the phone itself.


From dgarcia at anew.com.ve  Tue Aug 21 23:13:46 2012
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Tue, 21 Aug 2012 14:43:46 -0430
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <5033C5B1.3070509@hw.ac.uk>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<503226CE.7040307@hw.ac.uk> <50323D6A.3020708@anew.com.ve>
	<2526639.vT3cT5qGKG@virtex> <5033C5B1.3070509@hw.ac.uk>
Message-ID: <5033DDEA.9050305@anew.com.ve>

No problem Xing,

Callcenter is not an easy task, it is a worl itself.

As your "agents" are IVR apps, and if you like mod_callcenter, you could 
just create a GUI to "control/set" the pseudo-status of your IVR apps. 
Also, if the agent is busy with a call, mod_callcenter just distribute 
calls to free agents.

Good luck with your project.

Regards,
Dario

On 8/21/2012 1:00 PM, x.liu wrote:
> Hi g & Dario,
>
> Many thanks for your suggestions and explanations! Very helpful!
>
> I should have made my scenarios clearer in the first place:
> My agents are IVR apps which have registered to FS and are always
> available unless they are answering calls.
>
>
> Cheers,
> Xing
>
>
> On 08/20/2012 05:32 PM, g wrote:
>> Thanks Dario. You gave the answer I would like to do.
>> Liu, You can avoid agents dialing extensions to login/logout/pause, but at
>> least they have to click on some web app to let the system knows they are
>> available.
>> Another option is to let the system considerthem always available even when
>> the softphones are'nt registered. If you set a correct distribution criteria,
>> the unregistered softphones are skipped in a round robin distribution, as the
>> agents were not logged in. But in this case, forget about statistics on drops,
>> presence, waiting time on queue etc.
>> g
>>
>> On Monday 20 August 2012 09:06:42 Saugort Dario Garcia Tovar wrote:
>>> mmm,
>>>
>>> You think that the agent is dialing an extension. It is not what is
>>> happening. The agent is dialing a number that invoke a mod_callcenter
>>> function: login/logout/not ready. As FS is like a pbx (and more than a
>>> pbx), it seems like an extension.
>>>
>>> Just, registrer to FS is not enough. What happen when the agent need to
>>> go the restroom, take coffee, couching, or take a break? Then, you will
>>> need to reproduce in the agent gui the funcion of login, logout and
>>> break... but again, what happen if your agent gui fail? your callcenter
>>> service will stop because your gui fail? Also, if you just use register
>>> signaling, you could face a wide sort of operational issues like, what
>>> happen if the agent left the position and left the their seats: then
>>> calls will be routed to empty seats!
>>>
>>> If you want to get the agent login/logout without the agent dial
>>> nothing... if I am not wrong, mod_callcenter offer functions to log and
>>> logout agent from fs console, so I think you can invoke them using
>>> execute() from lua or your script.
>>>
>>> On 8/20/2012 7:30 AM, x.liu wrote:
>>>> Hi g,
>>>>
>>>> Thanks for your response!
>>>>
>>>> My situation is that I don't want the agent to dial the extension to be
>>>> able to accept calls.
>>>> The agents (e.g. a FS softphone client, or a Java application) just need
>>>> to register to FS as SIP clients.
>>>>
>>>> And I've not figured out what commands I should send if I use extension
>>>> "^agent-login$" and FS socket.
>>>>
>>>> So I guess I may be better not to use mod_callcenter for routing calls
>>>> to different extensions.
>>>> just to use FS ESL and my own logics to bridge the calls.
>>>>
>>>> I was curious about the reply "You don't need agent to dial anything to
>>>> get the call from queue... "
>>>> and I thought my question is relevant to the thread, so I asked
>>>> questions in this thread.
>>>> Apologies for that. I could have opened a different thread for the
>>>> questions :-)
>>>>
>>>> Cheers,
>>>> Xing
>>>>
>>>> On 08/20/2012 11:53 AM, g wrote:
>>>>> You can log an agent into a queue by a web application in which a
>>>>> button click handles commands via the FS socket, or just let the
>>>>> agent dial an extension (i.e. 999+itssoftphonenumber) where dialplan
>>>>> sets that agent in the correct state.
>>>>> ie (replace agent_login regex into something like 999201 if your agent
>>>>> is 201)<extension name="agent_login">
>>>>>
>>>>>       <condition field="destination_number"
>>>>>       expression="^agent-login$">
>>>>>
>>>>>         <action application="set"
>>>>>         data="res=${callcenter_config(agent set status>>
>>>>> ${caller_id_number}@${domain_name} 'Available')}" />
>>>>>
>>>>>         <action application="answer" data=""/>
>>>>>         <action application="sleep" data="500"/>
>>>>>         <action application="playback"
>>>>>         data="ivr/ivr-you_are_now_logged_in.wav"/>  <action
>>>>>         application="hangup" data=""/>
>>>>>
>>>>>       </condition>
>>>>>
>>>>> </extension>
>>>>>
>>>>> <extension name="agent_logoff">
>>>>>
>>>>>       <condition field="destination_number"
>>>>>       expression="^agent-logoff$">
>>>>>
>>>>>         <action application="set"
>>>>>         data="res=${callcenter_config(agent set status>>
>>>>> ${caller_id_number}@${domain_name} 'Logged Out')}" />
>>>>>
>>>>>         <action application="answer" data=""/>
>>>>>         <action application="sleep" data="500"/>
>>>>>         <action application="playback"
>>>>>         data="ivr/ivr-you_are_now_logged_out.wav"/>  <action
>>>>>         application="hangup" data=""/>
>>>>>
>>>>>       </condition>
>>>>>
>>>>> </extension>
>>>>>
>>>>> The important thing is tha the agent is, somehow, putted in
>>>>> "available"
>>>>> status. You can also do it via fs_cli, but now I can't remeber the
>>>>> exact
>>>>> syntax for that. Help from cli should clarify.
>>>>>
>>>>> At this point, the agent should result in "waiting" state
>>>>> Of course, you should before declared that agent as member of a tier
>>>>> or, at least, of a queue.
>>>>>
>>>>> Hope it helps
>>>>> g
>>>>>
>>>>> On Friday 17 August 2012 14:48:04 x.liu wrote:
>>>>>> I am also trying the mod_callcenter. What I want is to route a call
>>>>>> to
>>>>>> one agent ( a registered FS client, e.g. a SIP softphone)
>>>>>> of a list of agents by a Random or RoundRobin policy.
>>>>>>
>>>>>> When I dial the extension which calls the callcenter app, I only
>>>>>> have
>>>>>> hold-on music. It does not route a call to a agent.
>>>>>> You said " You don't need agent to dial anything to get the call
>>>>>> from
>>>>>> queue. ",
>>>>>> I am wondering how I could let an agent log into the queue?
>>>>>>
>>>>>> Thanks!
>>>>>>
>>>>>> Xing
>>>>>>
>>>>>> On 08/17/2012 01:56 PM, g wrote:
>>>>>>> Me too, I'm often dealing with callcenters (inbound+outbound). I
>>>>>>> found
>>>>>>> in
>>>>>>> Freeswitch the easyest solution to manage queue and agents. I'm
>>>>>>> not a
>>>>>>> programmer, but I have experience with Asterisk, Freeswitch and
>>>>>>> the
>>>>>>> whole
>>>>>>> workflow of complex callcenters.
>>>>>>> My suggestion too is to use mod_callcenter. Easy, fast, powerful.
>>>>>>> You
>>>>>>> don't need agent to dial anything to get the call from queue.
>>>>>>> In my opinion, you can use that module or, for more complex things
>>>>>>> (ie. outbound predictive) you can use the socket manager to
>>>>>>> perform  call placement, call transfer, conference with
>>>>>>> thirdy-part recording  IVR etc.
>>>>>>> I feel the best solution, also if it will require some effort for
>>>>>>> a Java programmer, is to do all that in php, which is light and
>>>>>>> fast also for older PC, able to run also with thin-client low
>>>>>>> memory low-cpu machines. If you can plan to turn to php and you
>>>>>>> don't have too short time, I can consider to help for the
>>>>>>> Freeswitch part.
>>>>>>> giuliano
>>>>>>>
>>>>>>> On Thursday 16 August 2012 17:10:33 Nitin Tomer wrote:
>>>>>>>> Hi,
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> I am working on a Contact Center solution. It will support mail,
>>>>>>>> chat
>>>>>>>> and call queries.
>>>>>>>>
>>>>>>>> The requirements are:
>>>>>>>>
>>>>>>>> 1. An end-customer calls, the call is handled by FreeSWITCH Auto
>>>>>>>> Attendant. 2. Customer is presented with a menu and makes
>>>>>>>> selection.
>>>>>>>> His call is put on hold and an entry is made in my system's
>>>>>>>> database
>>>>>>>> for incoming queries. 3. These queries are shown to agents
>>>>>>>> handling
>>>>>>>> calls.
>>>>>>>> 4. And agent clicks on a query, he is shown an extension where
>>>>>>>> call is parked. He dials that and is connected to the customer.
>>>>>>>>
>>>>>>>> 5. He talks to the customer and resolve his queries.
>>>>>>>>
>>>>>>>> Please guide me on how can do it. The application will be
>>>>>>>> written in
>>>>>>>> Java.
>>>>>>>>
>>>>>>>> I am an experienced programmers in Java/J2EE; but doesn't have
>>>>>>>> much
>>>>>>>> knowledge of VoIP/FreeSWITCH. I've configured FreeSWITCH on
>>>>>>>> Ubuntu
>>>>>>>> 12.04; and have installed X-Lite Softphones on two Windows
>>>>>>>> machines.
>>>>>>>> I configured these phones to work with FreeSWITCH and they are
>>>>>>>> working fine.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> I think that I'd have to use Valet Parking to park customer's
>>>>>>>> call on
>>>>>>>> an
>>>>>>>> extension, then pass that extension to agent who will dial it
>>>>>>>> and will be connected to the customer.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Please tell me whether this approach will work? And how should I
>>>>>>>> go
>>>>>>>> about it.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> All help would be much appreciated.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Regards
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Nitin
>>>>>>> __________________________________________________________________
>>>>>>> ______ _
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>> ______________________________________________________________________
>>>>> ___ Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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-use
>>>>> rs
>>>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
>


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

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Tel?fono: +58 212 9081842
Cel: +58 412 2221515
dgarcia at anew.com.ve
http://www.anew.com.ve
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From sat at calgaryit.com  Tue Aug 21 23:33:45 2012
From: sat at calgaryit.com (George Sapak)
Date: Tue, 21 Aug 2012 13:33:45 -0600 (MDT)
Subject: [Freeswitch-users] FreeBSD 8.3  Compile
In-Reply-To: <1792405498.66305.1345577564712.JavaMail.root@server3>
Message-ID: <575099670.66311.1345577625045.JavaMail.root@server3>

Anyone had luck compiling lately, I keep getting libjpeg error although I followed the wiki and installed tiff and jpeg.

Thanks, George




From nickolayr at gmail.com  Tue Aug 21 23:39:08 2012
From: nickolayr at gmail.com (Nikolay Rogoshchenkov)
Date: Tue, 21 Aug 2012 15:39:08 -0400
Subject: [Freeswitch-users] FreeBSD 8.3 Compile
In-Reply-To: <575099670.66311.1345577625045.JavaMail.root@server3>
References: <1792405498.66305.1345577564712.JavaMail.root@server3>
	<575099670.66311.1345577625045.JavaMail.root@server3>
Message-ID: <CAAQCs32BWvpyRg=6Yu6gWfvPo4wuDon+fZPsL_OoThcWRVmK7w@mail.gmail.com>

I'm this strange man that have to run FS at FreeBSD.
I have it on 8.2-RELEASE.

Are you sure that you have *libjpeg* on your system?

>ls /usr/local/lib/ | grep jpeg ?

--
Rogoshchenkov Nikolay


On Tue, Aug 21, 2012 at 3:33 PM, George Sapak <sat at calgaryit.com> wrote:

> Anyone had luck compiling lately, I keep getting libjpeg error although I
> followed the wiki and installed tiff and jpeg.
>
> Thanks, George
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sat at calgaryit.com  Tue Aug 21 23:45:22 2012
From: sat at calgaryit.com (George Sapak)
Date: Tue, 21 Aug 2012 13:45:22 -0600 (MDT)
Subject: [Freeswitch-users] FreeBSD 8.3 Compile
In-Reply-To: <CAAQCs32BWvpyRg=6Yu6gWfvPo4wuDon+fZPsL_OoThcWRVmK7w@mail.gmail.com>
Message-ID: <302013047.66423.1345578322329.JavaMail.root@server3>

That is what I get:

libjpeg.a
libjpeg.la
libjpeg.so
libjpeg.so.11

Thanks, George

----- Original Message -----
From: "Nikolay Rogoshchenkov" <nickolayr at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 21, 2012 1:39:08 PM
Subject: Re: [Freeswitch-users] FreeBSD 8.3 Compile



I'm this strange man that have to run FS at FreeBSD. 
I have it on 8.2-RELEASE. 


Are you sure that you have libjpeg on your system? 


>ls /usr/local/lib/ | grep jpeg ? 
-- 
Rogoshchenkov Nikolay 



On Tue, Aug 21, 2012 at 3:33 PM, George Sapak < sat at calgaryit.com > wrote: 


Anyone had luck compiling lately, I keep getting libjpeg error although I followed the wiki and installed tiff and jpeg. 

Thanks, George 



_________________________________________________________________________ 
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org 
http://www.freeswitchsolutions.com 

 
 

Official FreeSWITCH Sites 
http://www.freeswitch.org 
http://wiki.freeswitch.org 
http://www.cluecon.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 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 nickolayr at gmail.com  Tue Aug 21 23:49:12 2012
From: nickolayr at gmail.com (Nikolay Rogoshchenkov)
Date: Tue, 21 Aug 2012 15:49:12 -0400
Subject: [Freeswitch-users] FreeBSD 8.3 Compile
In-Reply-To: <302013047.66423.1345578322329.JavaMail.root@server3>
References: <CAAQCs32BWvpyRg=6Yu6gWfvPo4wuDon+fZPsL_OoThcWRVmK7w@mail.gmail.com>
	<302013047.66423.1345578322329.JavaMail.root@server3>
Message-ID: <CAAQCs33a5hyvkHRv1OSvBMcm_bpXkZt5b-AB4O-LPcsfcf96ag@mail.gmail.com>

try this before you start ./configure

setenv LDFLAGS -L/usr/local/lib
setenv CPPFLAGS -I/usr/local/include

--
Rogoshchenkov Nikolay


On Tue, Aug 21, 2012 at 3:45 PM, George Sapak <sat at calgaryit.com> wrote:

> That is what I get:
>
> libjpeg.a
> libjpeg.la
> libjpeg.so
> libjpeg.so.11
>
> Thanks, George
>
> ----- Original Message -----
> From: "Nikolay Rogoshchenkov" <nickolayr at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, August 21, 2012 1:39:08 PM
> Subject: Re: [Freeswitch-users] FreeBSD 8.3 Compile
>
>
>
> I'm this strange man that have to run FS at FreeBSD.
> I have it on 8.2-RELEASE.
>
>
> Are you sure that you have libjpeg on your system?
>
>
> >ls /usr/local/lib/ | grep jpeg ?
> --
> Rogoshchenkov Nikolay
>
>
>
> On Tue, Aug 21, 2012 at 3:33 PM, George Sapak < sat at calgaryit.com > wrote:
>
>
> Anyone had luck compiling lately, I keep getting libjpeg error although I
> followed the wiki and installed tiff and jpeg.
>
> Thanks, George
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sat at calgaryit.com  Tue Aug 21 23:59:36 2012
From: sat at calgaryit.com (George Sapak)
Date: Tue, 21 Aug 2012 13:59:36 -0600 (MDT)
Subject: [Freeswitch-users] FreeBSD 8.3 Compile
In-Reply-To: <CAAQCs33a5hyvkHRv1OSvBMcm_bpXkZt5b-AB4O-LPcsfcf96ag@mail.gmail.com>
Message-ID: <1439754696.66520.1345579176274.JavaMail.root@server3>

that worked, holy crap, its in the wiki but I thought that it olny applied to the unixODBC

Thank You very much, George

----- Original Message -----
From: "Nikolay Rogoshchenkov" <nickolayr at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 21, 2012 1:49:12 PM
Subject: Re: [Freeswitch-users] FreeBSD 8.3 Compile


try this before you start ./configure 


setenv LDFLAGS -L/usr/local/lib 
setenv CPPFLAGS -I/usr/local/include 

-- 
Rogoshchenkov Nikolay 



On Tue, Aug 21, 2012 at 3:45 PM, George Sapak < sat at calgaryit.com > wrote: 


That is what I get: 

libjpeg.a 
libjpeg.la 
libjpeg.so 
libjpeg.so.11 

Thanks, George 



----- Original Message ----- 
From: "Nikolay Rogoshchenkov" < nickolayr at gmail.com > 
To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > 
Sent: Tuesday, August 21, 2012 1:39:08 PM 
Subject: Re: [Freeswitch-users] FreeBSD 8.3 Compile 



I'm this strange man that have to run FS at FreeBSD. 
I have it on 8.2-RELEASE. 


Are you sure that you have libjpeg on your system? 


>ls /usr/local/lib/ | grep jpeg ? 
-- 
Rogoshchenkov Nikolay 



On Tue, Aug 21, 2012 at 3:33 PM, George Sapak < sat at calgaryit.com > wrote: 


Anyone had luck compiling lately, I keep getting libjpeg error although I followed the wiki and installed tiff and jpeg. 

Thanks, George 



_________________________________________________________________________ 
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org 
http://www.freeswitchsolutions.com 

 
 

Official FreeSWITCH Sites 
http://www.freeswitch.org 
http://wiki.freeswitch.org 
http://www.cluecon.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 


_________________________________________________________________________ 
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org 
http://www.freeswitchsolutions.com 

 
 

Official FreeSWITCH Sites 
http://www.freeswitch.org 
http://wiki.freeswitch.org 
http://www.cluecon.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 

_________________________________________________________________________ 
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org 
http://www.freeswitchsolutions.com 

 
 

Official FreeSWITCH Sites 
http://www.freeswitch.org 
http://wiki.freeswitch.org 
http://www.cluecon.com 

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From nickolayr at gmail.com  Wed Aug 22 00:05:20 2012
From: nickolayr at gmail.com (Nikolay Rogoshchenkov)
Date: Tue, 21 Aug 2012 16:05:20 -0400
Subject: [Freeswitch-users] FreeBSD 8.3 Compile
In-Reply-To: <1439754696.66520.1345579176274.JavaMail.root@server3>
References: <CAAQCs33a5hyvkHRv1OSvBMcm_bpXkZt5b-AB4O-LPcsfcf96ag@mail.gmail.com>
	<1439754696.66520.1345579176274.JavaMail.root@server3>
Message-ID: <CAAQCs32qXKM0Gdh=P=hweLZvfnpoRVgb5S9W7-WVMAfyB8zerA@mail.gmail.com>

Yes, it was added by *Alexander Sedov* (thank you!*)*, as I know from my
case FS-4205.
I think we need to clarify this in FreeBSD section.

--
Rogoshchenkov Nikolay


On Tue, Aug 21, 2012 at 3:59 PM, George Sapak <sat at calgaryit.com> wrote:

> that worked, holy crap, its in the wiki but I thought that it olny applied
> to the unixODBC
>
> Thank You very much, George
>
> ----- Original Message -----
> From: "Nikolay Rogoshchenkov" <nickolayr at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, August 21, 2012 1:49:12 PM
> Subject: Re: [Freeswitch-users] FreeBSD 8.3 Compile
>
>
> try this before you start ./configure
>
>
> setenv LDFLAGS -L/usr/local/lib
> setenv CPPFLAGS -I/usr/local/include
>
> --
> Rogoshchenkov Nikolay
>
>
>
> On Tue, Aug 21, 2012 at 3:45 PM, George Sapak < sat at calgaryit.com > wrote:
>
>
> That is what I get:
>
> libjpeg.a
> libjpeg.la
> libjpeg.so
> libjpeg.so.11
>
> Thanks, George
>
>
>
> ----- Original Message -----
> From: "Nikolay Rogoshchenkov" < nickolayr at gmail.com >
> To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org >
> Sent: Tuesday, August 21, 2012 1:39:08 PM
> Subject: Re: [Freeswitch-users] FreeBSD 8.3 Compile
>
>
>
> I'm this strange man that have to run FS at FreeBSD.
> I have it on 8.2-RELEASE.
>
>
> Are you sure that you have libjpeg on your system?
>
>
> >ls /usr/local/lib/ | grep jpeg ?
> --
> Rogoshchenkov Nikolay
>
>
>
> On Tue, Aug 21, 2012 at 3:33 PM, George Sapak < sat at calgaryit.com > wrote:
>
>
> Anyone had luck compiling lately, I keep getting libjpeg error although I
> followed the wiki and installed tiff and jpeg.
>
> Thanks, George
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mike at jerris.com  Wed Aug 22 00:16:16 2012
From: mike at jerris.com (Michael Jerris)
Date: Tue, 21 Aug 2012 16:16:16 -0400
Subject: [Freeswitch-users] FreeSWITCH and Pulse Audio
In-Reply-To: <k0usvt$h1r$1@ger.gmane.org>
References: <k0usvt$h1r$1@ger.gmane.org>
Message-ID: <08E8616A-92B7-464C-A5BD-6E87DC4ACEBF@jerris.com>

I would not use mod_alsa.  Is was added just for one device with very broken audio libraries and is not really maintained.  I would try out FreeSWITCH 1.2 and see how it works and report back.

Mike


On Aug 20, 2012, at 10:49 PM, Jason White <jason at jasonjgw.net> wrote:

> What, if any, is the right way to set up FreeSWITCH as a soft-phone on a
> system with Pulse Audio handling the audio device?
> 
> I found a bug on this topic that was never fixed, so I'm wondering what the
> solution now is and in which direction development is heading.
> 
> In the admittedly old version of FreeSWITCH that I'm running, mod_portaudio
> fails to find the device which is being handled by Pulse Audio as the default
> (it simply doesn't appear in the output of pa devlist).
> 
> I know there is also an Alsa module for FreeSWITCH, which used to be rather
> limited in functionality and not a substitute for mod_portaudio.
> 
> Thus my basic question is where FreeSWITCH is heading with regard to audio I/O
> under Linux. Is there anything interesting in the pipeline?
> 
> 



From mike at jerris.com  Wed Aug 22 00:17:34 2012
From: mike at jerris.com (Michael Jerris)
Date: Tue, 21 Aug 2012 16:17:34 -0400
Subject: [Freeswitch-users] No ringback with Lua Call Control
In-Reply-To: <1345545115.40493.YahooMailNeo@web132306.mail.ird.yahoo.com>
References: <1345545115.40493.YahooMailNeo@web132306.mail.ird.yahoo.com>
Message-ID: <E810704B-C802-4E57-AB6F-06BC34C70CC0@jerris.com>

You can do all this with a single originate command, why would you use lua for this in the first place?

Mike

On Aug 21, 2012, at 6:31 AM, Chris B. Ware <chrisbware at yahoo.it> wrote:

> Hi,
> 
> I need a small lua script to originate two extension calls  and bridge them.
> This is my script (based on lua wiki example):
> 
> local callee = argv[1]
> local called = argv[2]
> local domain = argv[3]
> 
> Adisp = "None";
> param="{ignore_early_media=true,ringback=\'%(1000,4000,425,425)\',origination_caller_id_number="
> 
> legA=freeswitch.Session(param..callee.."}user/"..called);
> while (Adisp~="ANSWER" and legA:ready()) do
>          Adisp  = legA:getVariable("endpoint_disposition");
>         os.execute("sleep 1")
> end -- waiting for A to answer
> if (Adisp=="ANSWER") then
>         legB=freeswitch.Session(param..called.."}user/"..callee);
>         freeswitch.bridge(legA,legB);
> end
> 
> The script works but there's no ringback, when A answers and B is ringing.
> When B answers, we can talk.
> 
> This is my log:
> http://pastebin.freeswitch.org/19745
> 
> 
> I know that something stupid is missing but I can't work out where's the error!
> Any help?

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From mike at jerris.com  Wed Aug 22 00:27:20 2012
From: mike at jerris.com (Michael Jerris)
Date: Tue, 21 Aug 2012 16:27:20 -0400
Subject: [Freeswitch-users] sip profile - accept-blind-auth vs
In-Reply-To: <5033CC06.2040605@gmail.com>
References: <5033CC06.2040605@gmail.com>
Message-ID: <6380AF79-A0F5-4A7C-A62F-4D5841C4848D@jerris.com>

auth-calls false means we won't challenge invite, accept-blind-auth means if auth headers are there, we ignore them.

Mike

On Aug 21, 2012, at 1:57 PM, Daniel-Constantin Mierla <miconda at gmail.com> wrote:

> Hello,
> 
> in the past I used to set:
> 
> <param name="auth-calls" value="false"/>
> 
> in the sip profile in order to skip user authentication for calls.
> 
> Lately I started to play a bit with 1.2 stable branch and seems that 
> setting auth-calls to false is no longer doing what I expected, calls 
> being challenged for user authentication.
> 
> Setting instead the accept-blind-auth to false got me what I wanted, like:
> 
>     <!-- accept any authentication without actually checking (not a 
> good feature for most people) -->
>     <param name="accept-blind-auth" value="true"/>
> 
> But from the comment (checked the wiki as well, but has the same text) 
> is a bit unclear what is the real purpose for it.
> 
> Isn't auth-calls=false supposed to accept calls without user 
> authentication anymore?
> 
> For this particular case, I play some announcements, like 'user not 
> available', and should work also for calls coming from outside. The 
> access is restricted by IP address ACL, allowing SIP traffic only from 
> my Kamailio instance.



From mike at jerris.com  Wed Aug 22 00:29:47 2012
From: mike at jerris.com (Michael Jerris)
Date: Tue, 21 Aug 2012 16:29:47 -0400
Subject: [Freeswitch-users] FreeBSD 8.3 Compile
In-Reply-To: <CAAQCs32qXKM0Gdh=P=hweLZvfnpoRVgb5S9W7-WVMAfyB8zerA@mail.gmail.com>
References: <CAAQCs33a5hyvkHRv1OSvBMcm_bpXkZt5b-AB4O-LPcsfcf96ag@mail.gmail.com>
	<1439754696.66520.1345579176274.JavaMail.root@server3>
	<CAAQCs32qXKM0Gdh=P=hweLZvfnpoRVgb5S9W7-WVMAfyB8zerA@mail.gmail.com>
Message-ID: <EAE4CA20-C986-4CDB-A1D5-13BC6E8671D7@jerris.com>

If anyone can provide patches to the build environment that will fix freebsd build (and not break others) this would be a help.  You can attach patches to the jira listed below.

Thanks
Mike

On Aug 21, 2012, at 4:05 PM, Nikolay Rogoshchenkov <nickolayr at gmail.com> wrote:

> Yes, it was added by Alexander Sedov (thank you!), as I know from my case FS-4205.
> I think we need to clarify this in FreeBSD section.
> 
> --
> Rogoshchenkov Nikolay
> 
> 
> On Tue, Aug 21, 2012 at 3:59 PM, George Sapak <sat at calgaryit.com> wrote:
> that worked, holy crap, its in the wiki but I thought that it olny applied to the unixODBC
> 
> Thank You very much, George
> 
> ----- Original Message -----
> From: "Nikolay Rogoshchenkov" <nickolayr at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, August 21, 2012 1:49:12 PM
> Subject: Re: [Freeswitch-users] FreeBSD 8.3 Compile
> 
> 
> try this before you start ./configure
> 
> 
> setenv LDFLAGS -L/usr/local/lib
> setenv CPPFLAGS -I/usr/local/include
> 
> --
> Rogoshchenkov Nikolay
> 
> 
> 
> On Tue, Aug 21, 2012 at 3:45 PM, George Sapak < sat at calgaryit.com > wrote:
> 
> 
> That is what I get:
> 
> libjpeg.a
> libjpeg.la
> libjpeg.so
> libjpeg.so.11
> 
> Thanks, George
> 
> 
> 
> ----- Original Message -----
> From: "Nikolay Rogoshchenkov" < nickolayr at gmail.com >
> To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org >
> Sent: Tuesday, August 21, 2012 1:39:08 PM
> Subject: Re: [Freeswitch-users] FreeBSD 8.3 Compile
> 
> 
> 
> I'm this strange man that have to run FS at FreeBSD.
> I have it on 8.2-RELEASE.
> 
> 
> Are you sure that you have libjpeg on your system?
> 
> 
> >ls /usr/local/lib/ | grep jpeg ?
> --
> Rogoshchenkov Nikolay
> 
> 
> 
> On Tue, Aug 21, 2012 at 3:33 PM, George Sapak < sat at calgaryit.com > wrote:
> 
> 
> Anyone had luck compiling lately, I keep getting libjpeg error although I followed the wiki and installed tiff and jpeg.
> 
> Thanks, George
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 miconda at gmail.com  Wed Aug 22 00:50:10 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Tue, 21 Aug 2012 22:50:10 +0200
Subject: [Freeswitch-users] sip profile - accept-blind-auth vs
In-Reply-To: <6380AF79-A0F5-4A7C-A62F-4D5841C4848D@jerris.com>
References: <5033CC06.2040605@gmail.com>
	<6380AF79-A0F5-4A7C-A62F-4D5841C4848D@jerris.com>
Message-ID: <5033F482.4070804@gmail.com>

Hi Mike,

On 8/21/12 10:27 PM, Michael Jerris wrote:
> auth-calls false means we won't challenge invite, accept-blind-auth means if auth headers are there, we ignore them.

it is what I expected from auth-calls (and worked like this in the 
past), but now even if set to false, the calls are challenged with 407 
reply for authentication. Only when I set accept-blind-auth to false 
there is no 407.

Overall, it gets me what I need, access being granted on IP acl, but I 
wanted to double check if such change in behaviour of auth-calls was 
done on purpose. I will review my changes comparing with the default 
configs to see if I modified other params that could result in this 
situation, although I think there is no other related parameter.

Cheers,
Daniel

>
> Mike
>
> On Aug 21, 2012, at 1:57 PM, Daniel-Constantin Mierla <miconda at gmail.com> wrote:
>
>> Hello,
>>
>> in the past I used to set:
>>
>> <param name="auth-calls" value="false"/>
>>
>> in the sip profile in order to skip user authentication for calls.
>>
>> Lately I started to play a bit with 1.2 stable branch and seems that
>> setting auth-calls to false is no longer doing what I expected, calls
>> being challenged for user authentication.
>>
>> Setting instead the accept-blind-auth to false got me what I wanted, like:
>>
>>      <!-- accept any authentication without actually checking (not a
>> good feature for most people) -->
>>      <param name="accept-blind-auth" value="true"/>
>>
>> But from the comment (checked the wiki as well, but has the same text)
>> is a bit unclear what is the real purpose for it.
>>
>> Isn't auth-calls=false supposed to accept calls without user
>> authentication anymore?
>>
>> For this particular case, I play some announcements, like 'user not
>> available', and should work also for calls coming from outside. The
>> access is restricted by IP address ACL, allowing SIP traffic only from
>> my Kamailio instance.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat



From mike at jerris.com  Wed Aug 22 01:16:06 2012
From: mike at jerris.com (Michael Jerris)
Date: Tue, 21 Aug 2012 17:16:06 -0400
Subject: [Freeswitch-users] sip profile - accept-blind-auth vs
In-Reply-To: <5033F482.4070804@gmail.com>
References: <5033CC06.2040605@gmail.com>
	<6380AF79-A0F5-4A7C-A62F-4D5841C4848D@jerris.com>
	<5033F482.4070804@gmail.com>
Message-ID: <74001ACB-E845-4B0C-AF94-FC284AC96E0B@jerris.com>

Sounds like a bug to me.  Could you file a jira on this?

Mike

On Aug 21, 2012, at 4:50 PM, Daniel-Constantin Mierla <miconda at gmail.com> wrote:

> Hi Mike,
> 
> On 8/21/12 10:27 PM, Michael Jerris wrote:
>> auth-calls false means we won't challenge invite, accept-blind-auth means if auth headers are there, we ignore them.
> 
> it is what I expected from auth-calls (and worked like this in the past), but now even if set to false, the calls are challenged with 407 reply for authentication. Only when I set accept-blind-auth to false there is no 407.
> 
> Overall, it gets me what I need, access being granted on IP acl, but I wanted to double check if such change in behaviour of auth-calls was done on purpose. I will review my changes comparing with the default configs to see if I modified other params that could result in this situation, although I think there is no other related parameter.
> 
> Cheers,
> Daniel
> 
>> 
>> Mike
>> 
>> On Aug 21, 2012, at 1:57 PM, Daniel-Constantin Mierla <miconda at gmail.com> wrote:
>> 
>>> Hello,
>>> 
>>> in the past I used to set:
>>> 
>>> <param name="auth-calls" value="false"/>
>>> 
>>> in the sip profile in order to skip user authentication for calls.
>>> 
>>> Lately I started to play a bit with 1.2 stable branch and seems that
>>> setting auth-calls to false is no longer doing what I expected, calls
>>> being challenged for user authentication.
>>> 
>>> Setting instead the accept-blind-auth to false got me what I wanted, like:
>>> 
>>>     <!-- accept any authentication without actually checking (not a
>>> good feature for most people) -->
>>>     <param name="accept-blind-auth" value="true"/>
>>> 
>>> But from the comment (checked the wiki as well, but has the same text)
>>> is a bit unclear what is the real purpose for it.
>>> 
>>> Isn't auth-calls=false supposed to accept calls without user
>>> authentication anymore?
>>> 
>>> For this particular case, I play some announcements, like 'user not
>>> available', and should work also for calls coming from outside. The
>>> access is restricted by IP address ACL, allowing SIP traffic only from
>>> my Kamailio instance.


From msc at freeswitch.org  Wed Aug 22 01:24:44 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 21 Aug 2012 14:24:44 -0700
Subject: [Freeswitch-users] FreeSWITCH Project Looking For Volunteers:
	Web Site Dev/Maint
In-Reply-To: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
Message-ID: <CAKzWOxVDK0dGOFOu7W86-7UDmEz+ksm-3=yhcOcgbw_trYHy4g@mail.gmail.com>

To all those who've pinged me: many thanks! In 24 hours I've received
information from 14 different individuals - this is excellent! "Many hands
makes light work" and all that. I will definitely be in touch. In the
meantime, if anyone else has Webby skills please don't hesitate to let me
know. Also, don't forget that you can also recruit people you know who
aren't part of the FreeSWITCH community.  :)

Thanks!

-Michael

On Mon, Aug 20, 2012 at 2:45 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hello all,
>
> I need to refresh my list of people who have Web dev skills and are
> willing to assist with things like Web site design, programming, and
> maintenance. If you have any of the following skills, or know someone who
> does and who is willing to donate a few hours, please email me offlist:
>
> HTML5
> CSS2/3
> Javascript/JQuery
> Drupal/PHP
> Wordpress/PHP
> Django/Python
> Web design/graphics
> Apache administration in LAMP environment
>
> Please note that not every single skill is currently in demand; we are
> trying to get a feel for who knows what and hopefully anticipate future
> needs.
>
> Thanks!
>
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From krice at freeswitch.org  Wed Aug 22 01:29:58 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 21 Aug 2012 16:29:58 -0500
Subject: [Freeswitch-users] FreeSWITCH Project Looking For Volunteers:
 Web Site Dev/Maint
In-Reply-To: <CAKzWOxVDK0dGOFOu7W86-7UDmEz+ksm-3=yhcOcgbw_trYHy4g@mail.gmail.com>
Message-ID: <CC596806.3C5B2%krice@freeswitch.org>

Hey also, Collins and other,

If any of you have Web Grafix/Layout skills contact me or collins... I have
a project in mind we could use a little possible help with

K


On 8/21/12 4:24 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> To all those who've pinged me: many thanks! In 24 hours I've received
> information from 14 different individuals - this is excellent! "Many hands
> makes light work" and all that. I will definitely be in touch. In the
> meantime, if anyone else has Webby skills please don't hesitate to let me
> know. Also, don't forget that you can also recruit people you know who aren't
> part of the FreeSWITCH community.? :)
> 
> Thanks!
> 
> -Michael
> 
> On Mon, Aug 20, 2012 at 2:45 PM, Michael Collins <msc at freeswitch.org> wrote:
>> Hello all,
>> 
>> I need to refresh my list of people who have Web dev skills and are willing
>> to assist with things like Web site design, programming, and maintenance. If
>> you have any of the following skills, or know someone who does and who is
>> willing to donate a few hours, please email me offlist:
>> 
>> HTML5
>> CSS2/3
>> Javascript/JQuery
>> Drupal/PHP
>> Wordpress/PHP
>> Django/Python
>> Web design/graphics
>> Apache administration in LAMP environment
>> 
>> Please note that not every single skill is currently in demand; we are trying
>> to get a feel for who knows what and hopefully anticipate future needs.
>> 
>> Thanks!
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From msc at freeswitch.org  Wed Aug 22 01:43:37 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 21 Aug 2012 14:43:37 -0700
Subject: [Freeswitch-users] sip profile - accept-blind-auth vs
In-Reply-To: <5033F482.4070804@gmail.com>
References: <5033CC06.2040605@gmail.com>
	<6380AF79-A0F5-4A7C-A62F-4D5841C4848D@jerris.com>
	<5033F482.4070804@gmail.com>
Message-ID: <CAKzWOxXMSmcr+aYwk30stnn15RxoEZUZwbTH9R=dN5pVWnyOBA@mail.gmail.com>

FWIW, I could not reproduce this behavior on v1.2.stable branch. When I set
auth-calls=false in the SIP profile and make an inbound call it just relies
on the ACL and that's it.

Miconda, what version of FS did you say you were running?

-MC

On Tue, Aug 21, 2012 at 1:50 PM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

> Hi Mike,
>
> On 8/21/12 10:27 PM, Michael Jerris wrote:
> > auth-calls false means we won't challenge invite, accept-blind-auth
> means if auth headers are there, we ignore them.
>
> it is what I expected from auth-calls (and worked like this in the
> past), but now even if set to false, the calls are challenged with 407
> reply for authentication. Only when I set accept-blind-auth to false
> there is no 407.
>
> Overall, it gets me what I need, access being granted on IP acl, but I
> wanted to double check if such change in behaviour of auth-calls was
> done on purpose. I will review my changes comparing with the default
> configs to see if I modified other params that could result in this
> situation, although I think there is no other related parameter.
>
> Cheers,
> Daniel
>
> >
> > Mike
> >
> > On Aug 21, 2012, at 1:57 PM, Daniel-Constantin Mierla <miconda at gmail.com>
> wrote:
> >
> >> Hello,
> >>
> >> in the past I used to set:
> >>
> >> <param name="auth-calls" value="false"/>
> >>
> >> in the sip profile in order to skip user authentication for calls.
> >>
> >> Lately I started to play a bit with 1.2 stable branch and seems that
> >> setting auth-calls to false is no longer doing what I expected, calls
> >> being challenged for user authentication.
> >>
> >> Setting instead the accept-blind-auth to false got me what I wanted,
> like:
> >>
> >>      <!-- accept any authentication without actually checking (not a
> >> good feature for most people) -->
> >>      <param name="accept-blind-auth" value="true"/>
> >>
> >> But from the comment (checked the wiki as well, but has the same text)
> >> is a bit unclear what is the real purpose for it.
> >>
> >> Isn't auth-calls=false supposed to accept calls without user
> >> authentication anymore?
> >>
> >> For this particular case, I play some announcements, like 'user not
> >> available', and should work also for calls coming from outside. The
> >> access is restricted by IP address ACL, allowing SIP traffic only from
> >> my Kamailio instance.
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
> http://asipto.com/u/kat
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jason.caulfield at intermetro.net  Wed Aug 22 01:35:52 2012
From: jason.caulfield at intermetro.net (Jason Caulfield)
Date: Tue, 21 Aug 2012 14:35:52 -0700
Subject: [Freeswitch-users] FreeSWITCH Question
Message-ID: <001f01cd7fe4$f1f29390$d5d7bab0$@caulfield@intermetro.net>

I am using FreeSWITCH in media bypass mode to decrease memory and cpu usage.


 

I notice that mod_sofia creates threads for each call leg to maintain
session context.

 

Do you know of a way to configure FreeSWITCH to use a table to maintain the
context when in media bypass mode to reduce the number of threads?

 

I am hoping that this will speed things up by reducing thread context
switching and reduce memory usage by decreasing memory allocation for each
thread.

 

Thanks for the help,

Jason

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From avi at avimarcus.net  Wed Aug 22 01:58:25 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 22 Aug 2012 00:58:25 +0300
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <5034019b.c456b60a.3fd2.ffffdc5cSMTPIN_ADDED@mx.google.com>
References: <5034019b.c456b60a.3fd2.ffffdc5cSMTPIN_ADDED@mx.google.com>
Message-ID: <CALNZuwmN+xfGkW0HqqDpXdJ4WaW5o59BcX-RiuKdNo18QH5D7g@mail.gmail.com>

Do you have such a large cps/concurrent volume that you're actually seeing
a performance hit?

-Avi


On Wed, Aug 22, 2012 at 12:35 AM, Jason Caulfield <
jason.caulfield at intermetro.net> wrote:

> I am using FreeSWITCH in media bypass mode to decrease memory and cpu
> usage.  ****
>
> ** **
>
> I notice that mod_sofia creates threads for each call leg to maintain
> session context.****
>
> ** **
>
> Do you know of a way to configure FreeSWITCH to use a table to maintain
> the context when in media bypass mode to reduce the number of threads?****
>
> ** **
>
> I am hoping that this will speed things up by reducing thread context
> switching and reduce memory usage by decreasing memory allocation for each
> thread.****
>
> ** **
>
> Thanks for the help,****
>
> Jason****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 krice at freeswitch.org  Wed Aug 22 02:19:20 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 21 Aug 2012 17:19:20 -0500
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <503401a3.679aec0a.5fe4.ffff9b6fSMTPIN_ADDED@mx.google.com>
Message-ID: <CC597398.3C5BD%krice@freeswitch.org>

Simple Answer No, FreeSWITCH is a B2BUA, the number of threads is 1 per call
leg, in bypass media, the thread is blocked for the most part and not
consuming anything until the hangup

At high call rates you?ll notice that most of the CPU is used by Sofia
threads...  Best way to reduce context switching is more cores or if you
have hyperthreading disabled, enabled it, you?ll notice a nice boost...

Tell Chris I said HI!
Ken


On 8/21/12 4:35 PM, "Jason Caulfield" <jason.caulfield at intermetro.net>
wrote:

> I am using FreeSWITCH in media bypass mode to decrease memory and cpu usage.
>  
> I notice that mod_sofia creates threads for each call leg to maintain session
> context.
>  
> Do you know of a way to configure FreeSWITCH to use a table to maintain the
> context when in media bypass mode to reduce the number of threads?
>  
> I am hoping that this will speed things up by reducing thread context
> switching and reduce memory usage by decreasing memory allocation for each
> thread.
>  
> Thanks for the help,
> Jason
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From jason.caulfield at intermetro.net  Wed Aug 22 02:38:30 2012
From: jason.caulfield at intermetro.net (Jason Caulfield)
Date: Tue, 21 Aug 2012 15:38:30 -0700
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <CALNZuwmN+xfGkW0HqqDpXdJ4WaW5o59BcX-RiuKdNo18QH5D7g@mail.gmail.com>
References: <5034019b.c456b60a.3fd2.ffffdc5cSMTPIN_ADDED@mx.google.com>
	<CALNZuwmN+xfGkW0HqqDpXdJ4WaW5o59BcX-RiuKdNo18QH5D7g@mail.gmail.com>
Message-ID: <004201cd7fed$b22ef150$168cd3f0$@caulfield@intermetro.net>

It is not so much the cps but the concurrent calls.

 

I am looking at maintaining a 30 cps with a 5 min. avg. call length which
results in a sustained 9000 concurrent calls.

 

This results in 85% mem. usage on a 4 GB machine, However CPU is at 25%
(total) on a dual core with hyper threading.

 

Current hardware does not support increase memory.

 

I am striving for about 75% cpu (appox. 90 cps).

 

Jason

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Tuesday, August 21, 2012 2:58 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSWITCH Question

 

Do you have such a large cps/concurrent volume that you're actually seeing a
performance hit?




-Avi

 

On Wed, Aug 22, 2012 at 12:35 AM, Jason Caulfield
<jason.caulfield at intermetro.net> wrote:

I am using FreeSWITCH in media bypass mode to decrease memory and cpu usage.


 

I notice that mod_sofia creates threads for each call leg to maintain
session context.

 

Do you know of a way to configure FreeSWITCH to use a table to maintain the
context when in media bypass mode to reduce the number of threads?

 

I am hoping that this will speed things up by reducing thread context
switching and reduce memory usage by decreasing memory allocation for each
thread.

 

Thanks for the help,

Jason


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 miconda at gmail.com  Wed Aug 22 02:33:56 2012
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Wed, 22 Aug 2012 00:33:56 +0200
Subject: [Freeswitch-users] sip profile - accept-blind-auth vs
In-Reply-To: <CAKzWOxXMSmcr+aYwk30stnn15RxoEZUZwbTH9R=dN5pVWnyOBA@mail.gmail.com>
References: <5033CC06.2040605@gmail.com>
	<6380AF79-A0F5-4A7C-A62F-4D5841C4848D@jerris.com>
	<5033F482.4070804@gmail.com>
	<CAKzWOxXMSmcr+aYwk30stnn15RxoEZUZwbTH9R=dN5pVWnyOBA@mail.gmail.com>
Message-ID: <50340CD4.9090006@gmail.com>

Hello,

the version is:

FreeSWITCH version: 1.2.1+git~20120816T172128Z~6dc9596bec (1.2.1; git at 
commit 6dc9596bec on Thu, 16 Aug 2012 17:21:28 Z)

Did few calls and got one more strange situation. So I commented the line:

<param name="accept-blind-auth" value="true"/>

and let:

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

Surprising, the calls are not authenticated.

But if I set:

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

I get 407 reply.

To summarize, I got:
1) auth-calls=false and accept-blind-auth=true => no 407 reply
2) auth-calls=false and accept-blind-auth commented => 407 reply
3) auth-calls=true and accept-blind-auth commented => no 407 reply

Looks like 3) is opposite than expected. Maybe it's too late in the 
night here, missing something obvious, I will try again tomorrow morning 
and I will fire a bug if it is really the case.

Cheers,
Daniel


On 8/21/12 11:43 PM, Michael Collins wrote:
> FWIW, I could not reproduce this behavior on v1.2.stable branch. When 
> I set auth-calls=false in the SIP profile and make an inbound call it 
> just relies on the ACL and that's it.
>
> Miconda, what version of FS did you say you were running?
>
> -MC
>
> On Tue, Aug 21, 2012 at 1:50 PM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hi Mike,
>
>     On 8/21/12 10:27 PM, Michael Jerris wrote:
>     > auth-calls false means we won't challenge invite,
>     accept-blind-auth means if auth headers are there, we ignore them.
>
>     it is what I expected from auth-calls (and worked like this in the
>     past), but now even if set to false, the calls are challenged with 407
>     reply for authentication. Only when I set accept-blind-auth to false
>     there is no 407.
>
>     Overall, it gets me what I need, access being granted on IP acl, but I
>     wanted to double check if such change in behaviour of auth-calls was
>     done on purpose. I will review my changes comparing with the default
>     configs to see if I modified other params that could result in this
>     situation, although I think there is no other related parameter.
>
>     Cheers,
>     Daniel
>
>     >
>     > Mike
>     >
>     > On Aug 21, 2012, at 1:57 PM, Daniel-Constantin Mierla
>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>     >
>     >> Hello,
>     >>
>     >> in the past I used to set:
>     >>
>     >> <param name="auth-calls" value="false"/>
>     >>
>     >> in the sip profile in order to skip user authentication for calls.
>     >>
>     >> Lately I started to play a bit with 1.2 stable branch and seems
>     that
>     >> setting auth-calls to false is no longer doing what I expected,
>     calls
>     >> being challenged for user authentication.
>     >>
>     >> Setting instead the accept-blind-auth to false got me what I
>     wanted, like:
>     >>
>     >>      <!-- accept any authentication without actually checking
>     (not a
>     >> good feature for most people) -->
>     >>      <param name="accept-blind-auth" value="true"/>
>     >>
>     >> But from the comment (checked the wiki as well, but has the
>     same text)
>     >> is a bit unclear what is the real purpose for it.
>     >>
>     >> Isn't auth-calls=false supposed to accept calls without user
>     >> authentication anymore?
>     >>
>     >> For this particular case, I play some announcements, like 'user not
>     >> available', and should work also for calls coming from outside. The
>     >> access is restricted by IP address ACL, allowing SIP traffic
>     only from
>     >> my Kamailio instance.
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     > http://www.freeswitchsolutions.com
>     >
>     > 
>     > 
>     >
>     > Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://wiki.freeswitch.org
>     > 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
>
>     --
>     Daniel-Constantin Mierla - http://www.asipto.com
>     http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -
>     http://www.linkedin.com/in/miconda
>     Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -
>     http://asipto.com/u/kat
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     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
>
>
>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

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From william.king at quentustech.com  Wed Aug 22 02:39:30 2012
From: william.king at quentustech.com (William King)
Date: Tue, 21 Aug 2012 15:39:30 -0700
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <004201cd7fed$b22ef150$168cd3f0$@caulfield@intermetro.net>
References: <5034019b.c456b60a.3fd2.ffffdc5cSMTPIN_ADDED@mx.google.com>
	<CALNZuwmN+xfGkW0HqqDpXdJ4WaW5o59BcX-RiuKdNo18QH5D7g@mail.gmail.com>
	<004201cd7fed$b22ef150$168cd3f0$@caulfield@intermetro.net>
Message-ID: <50340E22.2070309@quentustech.com>

Mind if I ask which specific cpu? and what OS?

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 08/21/2012 03:38 PM, Jason Caulfield wrote:
>
> It is not so much the cps but the concurrent calls.
>
> I am looking at maintaining a 30 cps with a 5 min. avg. call length 
> which results in a sustained 9000 concurrent calls.
>
> This results in 85% mem. usage on a 4 GB machine, However CPU is at 
> 25% (total) on a dual core with hyper threading.
>
> Current hardware does not support increase memory.
>
> I am striving for about 75% cpu (appox. 90 cps).
>
> Jason
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Avi Marcus
> *Sent:* Tuesday, August 21, 2012 2:58 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] FreeSWITCH Question
>
> Do you have such a large cps/concurrent volume that you're actually 
> seeing a performance hit?
>
>
> -Avi
>
> On Wed, Aug 22, 2012 at 12:35 AM, Jason Caulfield 
> <jason.caulfield at intermetro.net 
> <mailto:jason.caulfield at intermetro.net>> wrote:
>
> I am using FreeSWITCH in media bypass mode to decrease memory and cpu 
> usage.
>
> I notice that mod_sofia creates threads for each call leg to maintain 
> session context.
>
> Do you know of a way to configure FreeSWITCH to use a table to 
> maintain the context when in media bypass mode to reduce the number of 
> threads?
>
> I am hoping that this will speed things up by reducing thread context 
> switching and reduce memory usage by decreasing memory allocation for 
> each thread.
>
> Thanks for the help,
>
> Jason
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jason.caulfield at intermetro.net  Wed Aug 22 03:07:21 2012
From: jason.caulfield at intermetro.net (Jason Caulfield)
Date: Tue, 21 Aug 2012 16:07:21 -0700
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <CC597398.3C5BD%krice@freeswitch.org>
References: <503401a3.679aec0a.5fe4.ffff9b6fSMTPIN_ADDED@mx.google.com>
	<CC597398.3C5BD%krice@freeswitch.org>
Message-ID: <005f01cd7ff1$b8d20570$2a761050$@caulfield@intermetro.net>

Will do . . . that is say HI.

 

Hyperthreading is already on.  

 

The CPU usage is not an issue, it is the memory. 

 

At 25% CPU I am at 85% MEM.

 

Besides, it takes milliseconds to go through the dialplan and a few more for
logging, and with an average call length of 5 min, 99% of the time each
thread is doing nothing but taking up 1/2 a MB of memory.

 

Adds up quick . . . seems like a waist.

 

Anyone else, have knowledge on config-ing sofia?

 

Jason

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Tuesday, August 21, 2012 3:19 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSWITCH Question

 

Simple Answer No, FreeSWITCH is a B2BUA, the number of threads is 1 per call
leg, in bypass media, the thread is blocked for the most part and not
consuming anything until the hangup

At high call rates you'll notice that most of the CPU is used by Sofia
threads...  Best way to reduce context switching is more cores or if you
have hyperthreading disabled, enabled it, you'll notice a nice boost...

Tell Chris I said HI!
Ken


On 8/21/12 4:35 PM, "Jason Caulfield" <jason.caulfield at intermetro.net>
wrote:

I am using FreeSWITCH in media bypass mode to decrease memory and cpu usage.

 
I notice that mod_sofia creates threads for each call leg to maintain
session context.
 
Do you know of a way to configure FreeSWITCH to use a table to maintain the
context when in media bypass mode to reduce the number of threads?
 
I am hoping that this will speed things up by reducing thread context
switching and reduce memory usage by decreasing memory allocation for each
thread.
 
Thanks for the help,
Jason

  _____  

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From mike at jerris.com  Wed Aug 22 03:21:02 2012
From: mike at jerris.com (Michael Jerris)
Date: Tue, 21 Aug 2012 19:21:02 -0400
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <5034015b.6a74b60a.6840.ffffdac4SMTPIN_ADDED@mx.google.com>
References: <5034015b.6a74b60a.6840.ffffdac4SMTPIN_ADDED@mx.google.com>
Message-ID: <06BF2271-1E69-4A32-9E63-920FB45A82CA@jerris.com>

There is no way to do this.

On Aug 21, 2012, at 5:35 PM, Jason Caulfield <jason.caulfield at intermetro.net> wrote:

> I am using FreeSWITCH in media bypass mode to decrease memory and cpu usage. 
>  
> I notice that mod_sofia creates threads for each call leg to maintain session context.
>  
> Do you know of a way to configure FreeSWITCH to use a table to maintain the context when in media bypass mode to reduce the number of threads?
>  
> I am hoping that this will speed things up by reducing thread context switching and reduce memory usage by decreasing memory allocation for each thread.

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From mitch.capper at gmail.com  Wed Aug 22 09:21:58 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 21 Aug 2012 22:21:58 -0700
Subject: [Freeswitch-users] FSClient Windows Softphone 1.2 Release
Message-ID: <CALn233pcDdUPTbYMYTa_10cRWePxBwEe5XWHiSn_7077Nvdqsw@mail.gmail.com>

Some fairly large changes in this release.
Please see the UPNP Nat comment at the end of this if you are using FS
and are going through NAT. In addition if you are using any 3rd party
or self developed plugins see the note at the end.  I will briefly
talk about it on todays conference call also.

*) N-Way calling with conference support (mute/deaf/kick/split/energy
& volume level control on a per caller basis)
*) 3 Panel configurable layout, you can now collapse the call or accounts panel
*) Direct SIP: dialing, you can now dial into the conference for
example without having to use an actual server
*) Theme support choose from one of 4 themes for the dialer (Default
theme changed to Steel from former Royal Blue so make sure to change
back if you prefer the original look!)
*) Major startup time improvements, from 30+ seconds to under 5
*) Caller ID sending fix
*) Call auto answer support
*) Includes new portaudio fix that would cause your ring device to
sometimes ring while you were in a call
*) Account status and currently default account now shown on dialpad panel also.
*) Transfer Aliases, you can now right click on the XFER button and
select who to transfer to (right click on it prior to  being in a call
to add aliases).  This relies on the contact plugin to support storing
them, if you have a plugin based on SimpleContactPluginBase you must
atleast add stubs for the few new XFER functions.
*) Based on the recently minted FreeSWITCH 1.2 :)


Screenshots and details on the new options can be found at:
http://wiki.freeswitch.org/wiki/FSClient

Thanks also Jo?o for the nice comment on the last release! Sorry I
missed it I posted the last update right before i left on a trip for
several days.

The speed improvements on startup are in part due to the removal of
the UPNP nat check on startup,  I do not believe most people rely on
upnp however if you do find with it disabled (its in options now) your
FSClient doesn't work please let me know!   All plugins must now meet
the file naming convention in order to be loaded.  This means they
must end in HeadsetPlugin.dll or ContactPlugin.dll or else FSClient
will no longer try to load them.  This is to reduce load times and
prevent excess code from getting loaded into the executable.


~Mitch


From alex at thewinelake.com  Wed Aug 22 09:47:44 2012
From: alex at thewinelake.com (Alex Lake)
Date: Wed, 22 Aug 2012 06:47:44 +0100
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <004201cd7fed$b22ef150$168cd3f0$@caulfield@intermetro.net>
References: <5034019b.c456b60a.3fd2.ffffdc5cSMTPIN_ADDED@mx.google.com>
	<CALNZuwmN+xfGkW0HqqDpXdJ4WaW5o59BcX-RiuKdNo18QH5D7g@mail.gmail.com>
	<004201cd7fed$b22ef150$168cd3f0$@caulfield@intermetro.net>
Message-ID: <35C0D459-4C78-476F-B516-56BD8687DE54@digitalmail.com>

You want to do 90cps on a machine that has just 4G RAM?!



On 21 Aug 2012, at 23:38, "Jason Caulfield" <jason.caulfield at intermetro.net> wrote:

> It is not so much the cps but the concurrent calls.
>  
> I am looking at maintaining a 30 cps with a 5 min. avg. call length which results in a sustained 9000 concurrent calls.
>  
> This results in 85% mem. usage on a 4 GB machine, However CPU is at 25% (total) on a dual core with hyper threading.
>  
> Current hardware does not support increase memory.
>  
> I am striving for about 75% cpu (appox. 90 cps).
>  
> Jason
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus
> Sent: Tuesday, August 21, 2012 2:58 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FreeSWITCH Question
>  
> Do you have such a large cps/concurrent volume that you're actually seeing a performance hit?
> 
> -Avi
>  
> 
> On Wed, Aug 22, 2012 at 12:35 AM, Jason Caulfield <jason.caulfield at intermetro.net> wrote:
> I am using FreeSWITCH in media bypass mode to decrease memory and cpu usage. 
>  
> I notice that mod_sofia creates threads for each call leg to maintain session context.
>  
> Do you know of a way to configure FreeSWITCH to use a table to maintain the context when in media bypass mode to reduce the number of threads?
>  
> I am hoping that this will speed things up by reducing thread context switching and reduce memory usage by decreasing memory allocation for each thread.
>  
> Thanks for the help,
> Jason
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Wed Aug 22 10:12:38 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 22 Aug 2012 06:12:38 +0000
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <004201cd7fed$b22ef150$168cd3f0$@caulfield@intermetro.net>
References: <5034019b.c456b60a.3fd2.ffffdc5cSMTPIN_ADDED@mx.google.com>
	<CALNZuwmN+xfGkW0HqqDpXdJ4WaW5o59BcX-RiuKdNo18QH5D7g@mail.gmail.com>,
	<004201cd7fed$b22ef150$168cd3f0$@caulfield@intermetro.net>
Message-ID: <CD9593DE-1F02-4135-AEDD-7ED7C6A5FB22@visionutveckling.se>

If your current hardware doesn't support more RAM - buy a new server. FS scales well, but it will need the memory and CPU to perform.

To redesign FS to not create a new thread per channel would be a major rewrite of the entire core.

Considering your requirements it sonds like Kamailio or OpenSIPS is a better alternative for you, especially since you don't want to hande media anyway.

/Peter

22 aug 2012 kl. 00:39 skrev "Jason Caulfield" <jason.caulfield at intermetro.net<mailto:jason.caulfield at intermetro.net>>:

It is not so much the cps but the concurrent calls.

I am looking at maintaining a 30 cps with a 5 min. avg. call length which results in a sustained 9000 concurrent calls.

This results in 85% mem. usage on a 4 GB machine, However CPU is at 25% (total) on a dual core with hyper threading.

Current hardware does not support increase memory.

I am striving for about 75% cpu (appox. 90 cps).

Jason


From: 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 Avi Marcus
Sent: Tuesday, August 21, 2012 2:58 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSWITCH Question

Do you have such a large cps/concurrent volume that you're actually seeing a performance hit?

-Avi

On Wed, Aug 22, 2012 at 12:35 AM, Jason Caulfield <jason.caulfield at intermetro.net<mailto:jason.caulfield at intermetro.net>> wrote:
I am using FreeSWITCH in media bypass mode to decrease memory and cpu usage.

I notice that mod_sofia creates threads for each call leg to maintain session context.

Do you know of a way to configure FreeSWITCH to use a table to maintain the context when in media bypass mode to reduce the number of threads?

I am hoping that this will speed things up by reducing thread context switching and reduce memory usage by decreasing memory allocation for each thread.

Thanks for the help,
Jason

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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:50340a7d32766392456629!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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:50340a7d32766392456629!


From ntomer at newgen.co.in  Wed Aug 22 14:25:31 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Wed, 22 Aug 2012 15:55:31 +0530
Subject: [Freeswitch-users] Invalid speech module [cepstral] error on
	Flite
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD073F6F2858@NY1-EXMB-01.ip-soft.net>
References: <007a01cd7f8b$e82d3170$b8879450$@co.in>
	<6A6B4C284AD15042B429EB9D904544AD073F6F2858@NY1-EXMB-01.ip-soft.net>
Message-ID: <012101cd8050$7591c980$60b55c80$@co.in>

Thanks Hector.

 

I did what you suggested and it worked.

 

Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hector
Geraldino
Sent: Tuesday, August 21, 2012 11:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Invalid speech module [cepstral] error on
Flite

 

Hi,

 

Do you have cepstral binaries?

Have you uncommented the mod_cepstral  entry on the modules.conf.xml file in
conf/autoload_config?

If your intention is to use flite, have you changed the value for the
tts-engine in the en.xml file (conf/lang/en/en.xml) ?

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin
Tomer
Sent: Tuesday, August 21, 2012 6:59 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Invalid speech module [cepstral] error on Flite

 

Hi,

 

I am trying to make tts work using flite on freeswitch (installed/configured
using this guide - http://wiki.freeswitch.org/wiki/Installation_Guide) on
Ubuntu 12.04.

 

I configured a sample IVR menu using this -
<http://wiki.freeswitch.org/wiki/Examples_ivrmenu_js>
http://wiki.freeswitch.org/wiki/Examples_ivrmenu_js

 

I created a file ivrmenu.js in usr/local/freeswitch/scripts/; and made the
following entry in usr/local/freeswitch/conf/dialplan/default.xml:
 
<extension name="1200">
        <condition field="destination_number" expression="^1200$">
               <action application="javascript" data="ivrmenu.js"/>
        </condition>
</extension>
 

I added the following macro entry in
usr/local/freeswitch/conf/lang/en/en.xml

 

<macro name="mainmenu">
       <input pattern="0">
           <match>
               <action function="speak-text" data="Press 0 for the
Operator."/>
           </match>
       </input>
   
       <input pattern="1">
           <match>
               <action function="speak-text" data="Press 1 for
Accounting."/>
           </match>
       </input>
   
       <input pattern="2">
           <match>
               <action function="speak-text" data="Press 2 for Sales."/>
           </match>
       </input>
   
       <input pattern="3">
           <match>
               <action function="speak-text" data="Press 3 for Support."/>
           </match>
       </input>
   
       <input pattern="4">
           <match>
               <action function="speak-text" data="Press 4 for The
Janitor."/>
           </match>
       </input>
   
       <input pattern="5">
           <match>
               <action function="speak-text" data="Press 5 for The Smoking
Lounge."/>
           </match>
       </input>
   
       <input pattern="6">
           <match>
               <action function="speak-text" data="Press 6 for the men's
bathroom."/>
           </match>
       </input>
   
       <input pattern="7">
           <match>
               <action function="speak-text" data="Press 7 for the women's
powder room."/>
           </match>
       </input>
   
       <input pattern="8">
           <match>
               <action function="speak-text" data="Press 8 for no special
reason."/>
           </match>
       </input>
   
       <input pattern="9">
           <match>
               <action function="speak-text" data="Press 9 for a good
time."/>
           </match>
       </input>
   
       <input pattern="\*">
           <match>
               <action function="speak-text" data="Press star for good
luck."/>
           </match>
       </input>
   
       <input pattern="\#">
           <match>
               <action function="speak-text" data="Press pound for
psychological counseling."/>
           </match>
       </input>
   </macro>
   
   <macro name="saydigits">
       <input pattern="(.*)">
           <match>
               <action function="speak-text" data="You pressed $1"/>
           </match>
       </input>
   </macro>

 

Now when I dial extension 1200, following error is printed on Freeswitch
console -

 

2012-08-21 15:02:16.257525 [NOTICE] switch_channel.c:941 New Channel
sofia/internal/1000 at 192.168.6.109 [187ff238-eb73-11e1-a3ef-ed1990e665f3]

2012-08-21 15:02:16.257525 [INFO] mod_dialplan_xml.c:485 Processing 1000
<1000>->1200 in context default

2012-08-21 15:02:16.297537 [NOTICE] sofia_glue.c:4176 Pre-Answer
sofia/internal/1000 at 192.168.6.109!

2012-08-21 15:02:16.317513 [NOTICE] mod_spidermonkey.c:2098 Channel
[sofia/internal/1000 at 192.168.6.109] has been answered

2012-08-21 15:02:16.341524 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]!

2012-08-21 15:02:16.341524 [ERR] switch_ivr_play_say.c:2466 Invalid TTS
module!

2012-08-21 15:02:21.625528 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]!

2012-08-21 15:02:21.625528 [ERR] switch_ivr_play_say.c:2466 Invalid TTS
module!

2012-08-21 15:02:21.737528 [NOTICE] switch_core_state_machine.c:249
sofia/internal/1000 at 192.168.6.109 has executed the last dialplan
instruction, hanging up.

2012-08-21 15:02:21.737528 [NOTICE] switch_core_state_machine.c:251 Hangup
sofia/internal/1000 at 192.168.6.109 [CS_EXECUTE] [NORMAL_CLEARING]

2012-08-21 15:02:21.917533 [NOTICE] switch_core_session.c:1447 Session 1
(sofia/internal/1000 at 192.168.6.109) Ended

2012-08-21 15:02:21.917533 [NOTICE] switch_core_session.c:1449 Close Channel
sofia/internal/1000 at 192.168.6.109 [CS_DESTROY]

2012-08-21 15:02:32.957527 [NOTICE] switch_channel.c:941 New Channel
sofia/internal/1000 at 192.168.6.109 [2274b6a2-eb73-11e1-a3f5-ed1990e665f3]

2012-08-21 15:02:32.957527 [INFO] mod_dialplan_xml.c:485 Processing 1000
<1000>->1200 in context default

2012-08-21 15:02:32.978655 [NOTICE] sofia_glue.c:4176 Pre-Answer
sofia/internal/1000 at 192.168.6.109!

2012-08-21 15:02:32.978655 [NOTICE] mod_spidermonkey.c:2098 Channel
[sofia/internal/1000 at 192.168.6.109] has been answered

2012-08-21 15:02:33.005538 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]!

2012-08-21 15:02:33.005538 [ERR] switch_ivr_play_say.c:2466 Invalid TTS
module!

2012-08-21 15:02:39.124530 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]!

2012-08-21 15:02:39.124530 [ERR] switch_ivr_play_say.c:2466 Invalid TTS
module!

2012-08-21 15:02:45.265517 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]!

2012-08-21 15:02:45.265517 [ERR] switch_ivr_play_say.c:2466 Invalid TTS
module!

2012-08-21 15:02:51.385575 [ERR] switch_core_speech.c:61 Invalid speech
module [cepstral]!

2012-08-21 15:02:51.385575 [ERR] switch_ivr_play_say.c:2466 Invalid TTS
module!

2012-08-21 15:02:51.497528 [NOTICE] switch_core_state_machine.c:249
sofia/internal/1000 at 192.168.6.109 has executed the last dialplan
instruction, hanging up.

2012-08-21 15:02:51.497528 [NOTICE] switch_core_state_machine.c:251 Hangup
sofia/internal/1000 at 192.168.6.109 [CS_EXECUTE] [NORMAL_CLEARING]

2012-08-21 15:02:51.497528 [NOTICE] switch_core_session.c:1447 Session 2
(sofia/internal/1000 at 192.168.6.109) Ended

2012-08-21 15:02:51.497528 [NOTICE] switch_core_session.c:1449 Close Channel
sofia/internal/1000 at 192.168.6.109 [CS_DESTROY]

 

What seems to be the problem here? Am I missing something? Please help.

 

Thanks

 

Nitin

 

 


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From sameer2k3t at gmail.com  Wed Aug 22 14:49:05 2012
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Wed, 22 Aug 2012 15:49:05 +0500
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAMjaK3DCzKEi7Gep9eZLhGs4FGix3edCDCxtymJXCwWHP9FSjA@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
	<CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
	<CAA2gtHBXf-RfZ4eFB1k7NfCaYccqU6uK78NnxxBL0C=dO2PPBA@mail.gmail.com>
	<CAMjaK3DCzKEi7Gep9eZLhGs4FGix3edCDCxtymJXCwWHP9FSjA@mail.gmail.com>
Message-ID: <CAA2gtHAcp_2+ch0-TOK-6XibCdmdFzfWOvJ9s2_7JKLjnf7MTA@mail.gmail.com>

Hi,

It says

option                  8690  0
usb_wwan                9960  1 option
usbserial              31372  2 option,usb_wwan

What should I do now to get it working ?

Thanks for replying

On Tue, Aug 21, 2012 at 2:55 PM, Dmitry Lysenko
<dvl36.ripe.nick at gmail.com>wrote:

> As root: lsmod | grep option
> Should be something like that:
> ---
> root at debian:~# lsmod | grep option
> option                 15351  6
> usb_wwan                8362  1 option
> usbserial              26535  14 usb_wwan,option
> usbcore               122921  10
> ehci_hcd,usb_storage,usbserial,gspca_main,usb_wwan,option,gspca_zc3xx,snd_usbmidi_lib,snd_usb_audio
> ---
> 2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>
>> I guess so, I have CentOS 6 minimal installed
>>
>> how can I identify if this is option driver ?
>>
>> Yes I saw this message on ttyUSB0
>>
>> I can not load gsmopen with ttyUSB2 or ttyUSB1
>>
>>
>> On Tue, Aug 21, 2012 at 2:21 PM, Dmitry Lysenko <
>> dvl36.ripe.nick at gmail.com> wrote:
>>
>>> Are you using "option" linux driver?
>>> I checked Huawei E1550, E153 and E173 modems on some systems and they
>>> have only 3 virtual serial ports.(using "option" driver)
>>> Also you can check manually with minicom.
>>> Usually ttyUSB0 is accepting  AT-commands, but will not reply with OK or
>>> ERROR, ttyUSB1 is used for audio, and ttyUSB2 is accepting AT-commands and
>>> replying  as normal serial modem.
>>>
>>>
>>> 2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>
>>>> It didn't for me, according to wvdial my modem is ttyUSB0 and ttyUSB3
>>>>
>>>> On Mon, Aug 20, 2012 at 1:43 PM, Dmitry Lysenko <
>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>
>>>>>
>>>>> Hello.
>>>>> Try:
>>>>>  <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>> It's worked for me.
>>>>>
>>>>> 2012/8/20 Sameer Khan <sameer2k3t at gmail.com>
>>>>>
>>>>>>  Hi,
>>>>>>
>>>>>> I can't get audio working with e1550
>>>>>>
>>>>>> I have this config :
>>>>>>     <interface id="1" name="gsm01">
>>>>>>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB3"/>
>>>>>>
>>>>>> I tried swaping 0 and 3 but having same results
>>>>>>
>>>>>> I am originating a call from softphone and answering it on cell phone
>>>>>> but softphone doesn't get notified of being answered and keeps ringing
>>>>>>
>>>>>> Please help
>>>>>>
>>>>>> Thanks
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sameer2k3t at gmail.com  Wed Aug 22 14:56:16 2012
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Wed, 22 Aug 2012 15:56:16 +0500
Subject: [Freeswitch-users] mod_gsmopen: ROAMING Network ERROR
In-Reply-To: <CALXCt0q+CZsOBL6Uv0_W9TMje+qZGb+rguej-cgMqRNQA9mo6w@mail.gmail.com>
References: <CAD=Wdqi_rbz3Z0XrYcS_CP29wSV94G8R7cJpHKRqZoeNorhmxw@mail.gmail.com>
	<CAMjaK3B-Y+nswvVnj49Jh_+sZCJKvcXiSmhJ7ksMWo6UXbGt9Q@mail.gmail.com>
	<CALXCt0q+CZsOBL6Uv0_W9TMje+qZGb+rguej-cgMqRNQA9mo6w@mail.gmail.com>
Message-ID: <CAA2gtHAgmAC2J=5dMjWj1o2gzRbZRiRdmnbGjyfhide3T1_mhA@mail.gmail.com>

Hi dear,

no problem.,

yes Dimitri has extensive knowledge on gsmopen.

I think he will sort out my problem.

Thanks

On Tue, Aug 21, 2012 at 11:07 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> hi Sameer,
> I'm in holidays.
>
> If problems persists, please contact me after Sept 10th.
>
> Dimitri, you're right. Btw, you seems very knowledgeable about gsmopen
> and related stuff. Please feel free to modify expand mod-gsmopen wiki
> page and/or send patches to jira. Thanks in advance Dimitri!
>
> -giovanni
>
> On 8/21/12, Dmitry Lysenko <dvl36.ripe.nick at gmail.com> wrote:
> > Hello.
> > I have the same "ROAMING Network" ALERTs, it's annoying, but everything
> > working OK.
> > Checked with  Huawei E1550.
> >
> > P.S. Are you sure that your Huawei E169  is voice enabled?
> >
> > 2012/8/21 Josep Maria Vi?olas <vinolasbcn at gmail.com>
> >
> >> I have a Huawei E169 with a SIM from a "virtual provider" that uses
> other
> >> provider's networks, so it is allways technically 'roaming'.
> >>
> >> With mod_gsmopen the problem is that throws an error about "ROAMING
> >> Network" and also an ALERT. I don't know if that is why it is not able
> to
> >> make calls. In that case, how can I tell the module that roaming is not
> a
> >> bad thing??
> >>
> >> This is freeswitch debug output when I load mod_gsmmobile:
> >> ------------------------------------------
> >>
> > ...
> >
> >> ------------------------------------------
> >> ** From extension 1000 I hear a calling tone even the Freeswitch log
> says
> >> that ATD+N? command failed.
> >>
> >> Anyone knows if the problem is the ROAMING Network error??
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 dvl36.ripe.nick at gmail.com  Wed Aug 22 16:01:29 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Wed, 22 Aug 2012 15:01:29 +0300
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAA2gtHAcp_2+ch0-TOK-6XibCdmdFzfWOvJ9s2_7JKLjnf7MTA@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
	<CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
	<CAA2gtHBXf-RfZ4eFB1k7NfCaYccqU6uK78NnxxBL0C=dO2PPBA@mail.gmail.com>
	<CAMjaK3DCzKEi7Gep9eZLhGs4FGix3edCDCxtymJXCwWHP9FSjA@mail.gmail.com>
	<CAA2gtHAcp_2+ch0-TOK-6XibCdmdFzfWOvJ9s2_7JKLjnf7MTA@mail.gmail.com>
Message-ID: <CAMjaK3D+3XQDwU3s8m-FUj0wK_vPjOir3n3Lrvfwq9dS3Je2WQ@mail.gmail.com>

Hi,

I have reproduced  such situation. It seems that controldevice_audio_name
is pointing to wrong /dev/ttyUSBx.
I suggest try:

 <param name="controldevice_name" value="/dev/ttyUSB0"/>
 <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>

if doesn't help try:
 <param name="controldevice_name" value="/dev/ttyUSB0"/>
 <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>

P.S. Are you sure modem voice feature enabled?

2012/8/22 Sameer Khan <sameer2k3t at gmail.com>

> Hi,
>
> It says
>
> option                  8690  0
> usb_wwan                9960  1 option
> usbserial              31372  2 option,usb_wwan
>
> What should I do now to get it working ?
>
> Thanks for replying
>
> On Tue, Aug 21, 2012 at 2:55 PM, Dmitry Lysenko <dvl36.ripe.nick at gmail.com
> > wrote:
>
>> As root: lsmod | grep option
>> Should be something like that:
>> ---
>> root at debian:~# lsmod | grep option
>> option                 15351  6
>> usb_wwan                8362  1 option
>> usbserial              26535  14 usb_wwan,option
>> usbcore               122921  10
>> ehci_hcd,usb_storage,usbserial,gspca_main,usb_wwan,option,gspca_zc3xx,snd_usbmidi_lib,snd_usb_audio
>> ---
>>  2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>
>>> I guess so, I have CentOS 6 minimal installed
>>>
>>> how can I identify if this is option driver ?
>>>
>>> Yes I saw this message on ttyUSB0
>>>
>>> I can not load gsmopen with ttyUSB2 or ttyUSB1
>>>
>>>
>>> On Tue, Aug 21, 2012 at 2:21 PM, Dmitry Lysenko <
>>> dvl36.ripe.nick at gmail.com> wrote:
>>>
>>>> Are you using "option" linux driver?
>>>> I checked Huawei E1550, E153 and E173 modems on some systems and they
>>>> have only 3 virtual serial ports.(using "option" driver)
>>>> Also you can check manually with minicom.
>>>> Usually ttyUSB0 is accepting  AT-commands, but will not reply with OK
>>>> or ERROR, ttyUSB1 is used for audio, and ttyUSB2 is accepting AT-commands
>>>> and replying  as normal serial modem.
>>>>
>>>>
>>>> 2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>>
>>>>> It didn't for me, according to wvdial my modem is ttyUSB0 and ttyUSB3
>>>>>
>>>>> On Mon, Aug 20, 2012 at 1:43 PM, Dmitry Lysenko <
>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>
>>>>>>
>>>>>> Hello.
>>>>>> Try:
>>>>>>  <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>>> It's worked for me.
>>>>>>
>>>>>> 2012/8/20 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>
>>>>>>>  Hi,
>>>>>>>
>>>>>>> I can't get audio working with e1550
>>>>>>>
>>>>>>> I have this config :
>>>>>>>     <interface id="1" name="gsm01">
>>>>>>>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB3"/>
>>>>>>>
>>>>>>> I tried swaping 0 and 3 but having same results
>>>>>>>
>>>>>>> I am originating a call from softphone and answering it on cell
>>>>>>> phone but softphone doesn't get notified of being answered and keeps
>>>>>>> ringing
>>>>>>>
>>>>>>> Please help
>>>>>>>
>>>>>>> Thanks
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
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From jeff at jefflenk.com  Wed Aug 22 16:36:44 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 22 Aug 2012 05:36:44 -0700 (PDT)
Subject: [Freeswitch-users] FSClient Windows Softphone 1.2 Release
In-Reply-To: <CALn233pcDdUPTbYMYTa_10cRWePxBwEe5XWHiSn_7077Nvdqsw@mail.gmail.com>
References: <CALn233pcDdUPTbYMYTa_10cRWePxBwEe5XWHiSn_7077Nvdqsw@mail.gmail.com>
Message-ID: <1345639004510-7582055.post@n2.nabble.com>

This is awesome! thanks Mitch. I haven't had much time to test it but I will
let you know more in the next weeks.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FSClient-Windows-Softphone-1-2-Release-tp7582048p7582055.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sameer2k3t at gmail.com  Wed Aug 22 16:39:23 2012
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Wed, 22 Aug 2012 17:39:23 +0500
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAMjaK3D+3XQDwU3s8m-FUj0wK_vPjOir3n3Lrvfwq9dS3Je2WQ@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
	<CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
	<CAA2gtHBXf-RfZ4eFB1k7NfCaYccqU6uK78NnxxBL0C=dO2PPBA@mail.gmail.com>
	<CAMjaK3DCzKEi7Gep9eZLhGs4FGix3edCDCxtymJXCwWHP9FSjA@mail.gmail.com>
	<CAA2gtHAcp_2+ch0-TOK-6XibCdmdFzfWOvJ9s2_7JKLjnf7MTA@mail.gmail.com>
	<CAMjaK3D+3XQDwU3s8m-FUj0wK_vPjOir3n3Lrvfwq9dS3Je2WQ@mail.gmail.com>
Message-ID: <CAA2gtHBHkmUWM6EyqP14RtoH-jLUWitUdHXkZYKAVp3e54eBmw@mail.gmail.com>

No luck

2012-08-22 17:35:38.337967 [ERR] gsmopen_protocol.cpp:3160 rev
ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
ttyUSB2, NOT open

2012-08-22 17:37:27.057967 [ERR] gsmopen_protocol.cpp:3160 rev
ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
ttyUSB1, NOT open

Yes It is enabled, I tested before on other box


On Wed, Aug 22, 2012 at 5:01 PM, Dmitry Lysenko
<dvl36.ripe.nick at gmail.com>wrote:

> Hi,
>
> I have reproduced  such situation. It seems that controldevice_audio_name
> is pointing to wrong /dev/ttyUSBx.
> I suggest try:
>
>  <param name="controldevice_name" value="/dev/ttyUSB0"/>
>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>
> if doesn't help try:
>  <param name="controldevice_name" value="/dev/ttyUSB0"/>
>  <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>
> P.S. Are you sure modem voice feature enabled?
>
> 2012/8/22 Sameer Khan <sameer2k3t at gmail.com>
>
>> Hi,
>>
>> It says
>>
>> option                  8690  0
>> usb_wwan                9960  1 option
>> usbserial              31372  2 option,usb_wwan
>>
>> What should I do now to get it working ?
>>
>> Thanks for replying
>>
>> On Tue, Aug 21, 2012 at 2:55 PM, Dmitry Lysenko <
>> dvl36.ripe.nick at gmail.com> wrote:
>>
>>> As root: lsmod | grep option
>>> Should be something like that:
>>> ---
>>> root at debian:~# lsmod | grep option
>>> option                 15351  6
>>> usb_wwan                8362  1 option
>>> usbserial              26535  14 usb_wwan,option
>>> usbcore               122921  10
>>> ehci_hcd,usb_storage,usbserial,gspca_main,usb_wwan,option,gspca_zc3xx,snd_usbmidi_lib,snd_usb_audio
>>> ---
>>>  2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>
>>>> I guess so, I have CentOS 6 minimal installed
>>>>
>>>> how can I identify if this is option driver ?
>>>>
>>>> Yes I saw this message on ttyUSB0
>>>>
>>>> I can not load gsmopen with ttyUSB2 or ttyUSB1
>>>>
>>>>
>>>> On Tue, Aug 21, 2012 at 2:21 PM, Dmitry Lysenko <
>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>
>>>>> Are you using "option" linux driver?
>>>>> I checked Huawei E1550, E153 and E173 modems on some systems and they
>>>>> have only 3 virtual serial ports.(using "option" driver)
>>>>> Also you can check manually with minicom.
>>>>> Usually ttyUSB0 is accepting  AT-commands, but will not reply with OK
>>>>> or ERROR, ttyUSB1 is used for audio, and ttyUSB2 is accepting AT-commands
>>>>> and replying  as normal serial modem.
>>>>>
>>>>>
>>>>> 2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>>>
>>>>>> It didn't for me, according to wvdial my modem is ttyUSB0 and ttyUSB3
>>>>>>
>>>>>> On Mon, Aug 20, 2012 at 1:43 PM, Dmitry Lysenko <
>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>
>>>>>>>
>>>>>>> Hello.
>>>>>>> Try:
>>>>>>>  <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>>>> It's worked for me.
>>>>>>>
>>>>>>> 2012/8/20 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>
>>>>>>>>  Hi,
>>>>>>>>
>>>>>>>> I can't get audio working with e1550
>>>>>>>>
>>>>>>>> I have this config :
>>>>>>>>     <interface id="1" name="gsm01">
>>>>>>>>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB3"/>
>>>>>>>>
>>>>>>>> I tried swaping 0 and 3 but having same results
>>>>>>>>
>>>>>>>> I am originating a call from softphone and answering it on cell
>>>>>>>> phone but softphone doesn't get notified of being answered and keeps
>>>>>>>> ringing
>>>>>>>>
>>>>>>>> Please help
>>>>>>>>
>>>>>>>> Thanks
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
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From sameer2k3t at gmail.com  Wed Aug 22 16:45:38 2012
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Wed, 22 Aug 2012 17:45:38 +0500
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAA2gtHBHkmUWM6EyqP14RtoH-jLUWitUdHXkZYKAVp3e54eBmw@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
	<CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
	<CAA2gtHBXf-RfZ4eFB1k7NfCaYccqU6uK78NnxxBL0C=dO2PPBA@mail.gmail.com>
	<CAMjaK3DCzKEi7Gep9eZLhGs4FGix3edCDCxtymJXCwWHP9FSjA@mail.gmail.com>
	<CAA2gtHAcp_2+ch0-TOK-6XibCdmdFzfWOvJ9s2_7JKLjnf7MTA@mail.gmail.com>
	<CAMjaK3D+3XQDwU3s8m-FUj0wK_vPjOir3n3Lrvfwq9dS3Je2WQ@mail.gmail.com>
	<CAA2gtHBHkmUWM6EyqP14RtoH-jLUWitUdHXkZYKAVp3e54eBmw@mail.gmail.com>
Message-ID: <CAA2gtHCpqRFB+d8xE5DWYbuC2U8uaWV_56VUzpSh7t2zzUvtHQ@mail.gmail.com>

this is wvdial output


Found a modem on /dev/ttyUSB0.
Modem configuration written to /etc/wvdial.conf.
ttyUSB0<Info>: Speed 9600; init "ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0"
ttyUSB3<Info>: Speed 9600; init "ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0"


On Wed, Aug 22, 2012 at 5:39 PM, Sameer Khan <sameer2k3t at gmail.com> wrote:

> No luck
>
> 2012-08-22 17:35:38.337967 [ERR] gsmopen_protocol.cpp:3160 rev
> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
> ttyUSB2, NOT open
>
> 2012-08-22 17:37:27.057967 [ERR] gsmopen_protocol.cpp:3160 rev
> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
> ttyUSB1, NOT open
>
> Yes It is enabled, I tested before on other box
>
>
> On Wed, Aug 22, 2012 at 5:01 PM, Dmitry Lysenko <dvl36.ripe.nick at gmail.com
> > wrote:
>
>> Hi,
>>
>> I have reproduced  such situation. It seems that controldevice_audio_name
>> is pointing to wrong /dev/ttyUSBx.
>> I suggest try:
>>
>>  <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>
>> if doesn't help try:
>>  <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>  <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>>
>> P.S. Are you sure modem voice feature enabled?
>>
>> 2012/8/22 Sameer Khan <sameer2k3t at gmail.com>
>>
>>> Hi,
>>>
>>> It says
>>>
>>> option                  8690  0
>>> usb_wwan                9960  1 option
>>> usbserial              31372  2 option,usb_wwan
>>>
>>> What should I do now to get it working ?
>>>
>>> Thanks for replying
>>>
>>> On Tue, Aug 21, 2012 at 2:55 PM, Dmitry Lysenko <
>>> dvl36.ripe.nick at gmail.com> wrote:
>>>
>>>> As root: lsmod | grep option
>>>> Should be something like that:
>>>> ---
>>>> root at debian:~# lsmod | grep option
>>>> option                 15351  6
>>>> usb_wwan                8362  1 option
>>>> usbserial              26535  14 usb_wwan,option
>>>> usbcore               122921  10
>>>> ehci_hcd,usb_storage,usbserial,gspca_main,usb_wwan,option,gspca_zc3xx,snd_usbmidi_lib,snd_usb_audio
>>>> ---
>>>>  2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>>
>>>>> I guess so, I have CentOS 6 minimal installed
>>>>>
>>>>> how can I identify if this is option driver ?
>>>>>
>>>>> Yes I saw this message on ttyUSB0
>>>>>
>>>>> I can not load gsmopen with ttyUSB2 or ttyUSB1
>>>>>
>>>>>
>>>>> On Tue, Aug 21, 2012 at 2:21 PM, Dmitry Lysenko <
>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>
>>>>>> Are you using "option" linux driver?
>>>>>> I checked Huawei E1550, E153 and E173 modems on some systems and they
>>>>>> have only 3 virtual serial ports.(using "option" driver)
>>>>>> Also you can check manually with minicom.
>>>>>> Usually ttyUSB0 is accepting  AT-commands, but will not reply with OK
>>>>>> or ERROR, ttyUSB1 is used for audio, and ttyUSB2 is accepting AT-commands
>>>>>> and replying  as normal serial modem.
>>>>>>
>>>>>>
>>>>>> 2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>
>>>>>>> It didn't for me, according to wvdial my modem is ttyUSB0 and ttyUSB3
>>>>>>>
>>>>>>> On Mon, Aug 20, 2012 at 1:43 PM, Dmitry Lysenko <
>>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>>
>>>>>>>>
>>>>>>>> Hello.
>>>>>>>> Try:
>>>>>>>>  <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>>>>> It's worked for me.
>>>>>>>>
>>>>>>>> 2012/8/20 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>>
>>>>>>>>>  Hi,
>>>>>>>>>
>>>>>>>>> I can't get audio working with e1550
>>>>>>>>>
>>>>>>>>> I have this config :
>>>>>>>>>     <interface id="1" name="gsm01">
>>>>>>>>>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB3"/>
>>>>>>>>>
>>>>>>>>> I tried swaping 0 and 3 but having same results
>>>>>>>>>
>>>>>>>>> I am originating a call from softphone and answering it on cell
>>>>>>>>> phone but softphone doesn't get notified of being answered and keeps
>>>>>>>>> ringing
>>>>>>>>>
>>>>>>>>> Please help
>>>>>>>>>
>>>>>>>>> Thanks
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 dvl36.ripe.nick at gmail.com  Wed Aug 22 17:03:55 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Wed, 22 Aug 2012 16:03:55 +0300
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAA2gtHCpqRFB+d8xE5DWYbuC2U8uaWV_56VUzpSh7t2zzUvtHQ@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
	<CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
	<CAA2gtHBXf-RfZ4eFB1k7NfCaYccqU6uK78NnxxBL0C=dO2PPBA@mail.gmail.com>
	<CAMjaK3DCzKEi7Gep9eZLhGs4FGix3edCDCxtymJXCwWHP9FSjA@mail.gmail.com>
	<CAA2gtHAcp_2+ch0-TOK-6XibCdmdFzfWOvJ9s2_7JKLjnf7MTA@mail.gmail.com>
	<CAMjaK3D+3XQDwU3s8m-FUj0wK_vPjOir3n3Lrvfwq9dS3Je2WQ@mail.gmail.com>
	<CAA2gtHBHkmUWM6EyqP14RtoH-jLUWitUdHXkZYKAVp3e54eBmw@mail.gmail.com>
	<CAA2gtHCpqRFB+d8xE5DWYbuC2U8uaWV_56VUzpSh7t2zzUvtHQ@mail.gmail.com>
Message-ID: <CAMjaK3A0aKF80nFCm-roRB1MC65RA3Rt-0QErx=6py8L4K7r_w@mail.gmail.com>

Maybe:
http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/087009.html
?

2012/8/22 Sameer Khan <sameer2k3t at gmail.com>

> this is wvdial output
>
>
> Found a modem on /dev/ttyUSB0.
> Modem configuration written to /etc/wvdial.conf.
> ttyUSB0<Info>: Speed 9600; init "ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0"
> ttyUSB3<Info>: Speed 9600; init "ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0"
>
>
> On Wed, Aug 22, 2012 at 5:39 PM, Sameer Khan <sameer2k3t at gmail.com> wrote:
>
>> No luck
>>
>> 2012-08-22 17:35:38.337967 [ERR] gsmopen_protocol.cpp:3160 rev
>> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
>> ttyUSB2, NOT open
>>
>> 2012-08-22 17:37:27.057967 [ERR] gsmopen_protocol.cpp:3160 rev
>> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
>> ttyUSB1, NOT open
>>
>> Yes It is enabled, I tested before on other box
>>
>>
>> On Wed, Aug 22, 2012 at 5:01 PM, Dmitry Lysenko <
>> dvl36.ripe.nick at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I have reproduced  such situation. It seems that
>>> controldevice_audio_name is pointing to wrong /dev/ttyUSBx.
>>> I suggest try:
>>>
>>>  <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>
>>> if doesn't help try:
>>>  <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>>>
>>> P.S. Are you sure modem voice feature enabled?
>>>
>>> 2012/8/22 Sameer Khan <sameer2k3t at gmail.com>
>>>
>>>> Hi,
>>>>
>>>> It says
>>>>
>>>> option                  8690  0
>>>> usb_wwan                9960  1 option
>>>> usbserial              31372  2 option,usb_wwan
>>>>
>>>> What should I do now to get it working ?
>>>>
>>>> Thanks for replying
>>>>
>>>> On Tue, Aug 21, 2012 at 2:55 PM, Dmitry Lysenko <
>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>
>>>>> As root: lsmod | grep option
>>>>> Should be something like that:
>>>>> ---
>>>>> root at debian:~# lsmod | grep option
>>>>> option                 15351  6
>>>>> usb_wwan                8362  1 option
>>>>> usbserial              26535  14 usb_wwan,option
>>>>> usbcore               122921  10
>>>>> ehci_hcd,usb_storage,usbserial,gspca_main,usb_wwan,option,gspca_zc3xx,snd_usbmidi_lib,snd_usb_audio
>>>>> ---
>>>>>  2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>>>
>>>>>> I guess so, I have CentOS 6 minimal installed
>>>>>>
>>>>>> how can I identify if this is option driver ?
>>>>>>
>>>>>> Yes I saw this message on ttyUSB0
>>>>>>
>>>>>> I can not load gsmopen with ttyUSB2 or ttyUSB1
>>>>>>
>>>>>>
>>>>>> On Tue, Aug 21, 2012 at 2:21 PM, Dmitry Lysenko <
>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>
>>>>>>> Are you using "option" linux driver?
>>>>>>> I checked Huawei E1550, E153 and E173 modems on some systems and
>>>>>>> they have only 3 virtual serial ports.(using "option" driver)
>>>>>>> Also you can check manually with minicom.
>>>>>>> Usually ttyUSB0 is accepting  AT-commands, but will not reply with
>>>>>>> OK or ERROR, ttyUSB1 is used for audio, and ttyUSB2 is accepting
>>>>>>> AT-commands and replying  as normal serial modem.
>>>>>>>
>>>>>>>
>>>>>>> 2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>
>>>>>>>> It didn't for me, according to wvdial my modem is ttyUSB0 and
>>>>>>>> ttyUSB3
>>>>>>>>
>>>>>>>> On Mon, Aug 20, 2012 at 1:43 PM, Dmitry Lysenko <
>>>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>>>
>>>>>>>>>
>>>>>>>>> Hello.
>>>>>>>>> Try:
>>>>>>>>>  <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>>>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>>>>>> It's worked for me.
>>>>>>>>>
>>>>>>>>> 2012/8/20 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>>>
>>>>>>>>>>  Hi,
>>>>>>>>>>
>>>>>>>>>> I can't get audio working with e1550
>>>>>>>>>>
>>>>>>>>>> I have this config :
>>>>>>>>>>     <interface id="1" name="gsm01">
>>>>>>>>>>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>>>>>>       <param name="controldevice_audio_name"
>>>>>>>>>> value="/dev/ttyUSB3"/>
>>>>>>>>>>
>>>>>>>>>> I tried swaping 0 and 3 but having same results
>>>>>>>>>>
>>>>>>>>>> I am originating a call from softphone and answering it on cell
>>>>>>>>>> phone but softphone doesn't get notified of being answered and keeps
>>>>>>>>>> ringing
>>>>>>>>>>
>>>>>>>>>> Please help
>>>>>>>>>>
>>>>>>>>>> Thanks
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 dvl36.ripe.nick at gmail.com  Wed Aug 22 17:35:23 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Wed, 22 Aug 2012 16:35:23 +0300
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAA2gtHBHkmUWM6EyqP14RtoH-jLUWitUdHXkZYKAVp3e54eBmw@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
	<CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
	<CAA2gtHBXf-RfZ4eFB1k7NfCaYccqU6uK78NnxxBL0C=dO2PPBA@mail.gmail.com>
	<CAMjaK3DCzKEi7Gep9eZLhGs4FGix3edCDCxtymJXCwWHP9FSjA@mail.gmail.com>
	<CAA2gtHAcp_2+ch0-TOK-6XibCdmdFzfWOvJ9s2_7JKLjnf7MTA@mail.gmail.com>
	<CAMjaK3D+3XQDwU3s8m-FUj0wK_vPjOir3n3Lrvfwq9dS3Je2WQ@mail.gmail.com>
	<CAA2gtHBHkmUWM6EyqP14RtoH-jLUWitUdHXkZYKAVp3e54eBmw@mail.gmail.com>
Message-ID: <CAMjaK3BfjTzYkfdGo59MGo4fKxm=v0wVR6Ro_MKqGt-fcdf_+w@mail.gmail.com>

2012/8/22 Sameer Khan <sameer2k3t at gmail.com>

> No luck
>
> 2012-08-22 17:35:38.337967 [ERR] gsmopen_protocol.cpp:3160 rev
> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
> ttyUSB2, NOT open
>

Sameer, should be */dev/*ttyUSB2


>From Mi Ke log:

2012-08-21 09:18:18.973501 [ERR] gsmopen_protocol.cpp:122 rev
ccae5cd|07bc7ba[(nil)|37 ][ERRORA 122 ][gsm00 ][-1, 0, 0] port
*/dev/ttyUSB0*, NOT open

And my log:

gsmopen_protocol.cpp:120 rev ccae5cd|07bc7ba[(nil)|37
][DEBUG_GSMOPEN  120  ][gsm01     ][-1, 0, 0] port */dev/ttyUSB2*,
SUCCESS open
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From sameer2k3t at gmail.com  Wed Aug 22 17:44:21 2012
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Wed, 22 Aug 2012 18:44:21 +0500
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAMjaK3A0aKF80nFCm-roRB1MC65RA3Rt-0QErx=6py8L4K7r_w@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
	<CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
	<CAA2gtHBXf-RfZ4eFB1k7NfCaYccqU6uK78NnxxBL0C=dO2PPBA@mail.gmail.com>
	<CAMjaK3DCzKEi7Gep9eZLhGs4FGix3edCDCxtymJXCwWHP9FSjA@mail.gmail.com>
	<CAA2gtHAcp_2+ch0-TOK-6XibCdmdFzfWOvJ9s2_7JKLjnf7MTA@mail.gmail.com>
	<CAMjaK3D+3XQDwU3s8m-FUj0wK_vPjOir3n3Lrvfwq9dS3Je2WQ@mail.gmail.com>
	<CAA2gtHBHkmUWM6EyqP14RtoH-jLUWitUdHXkZYKAVp3e54eBmw@mail.gmail.com>
	<CAA2gtHCpqRFB+d8xE5DWYbuC2U8uaWV_56VUzpSh7t2zzUvtHQ@mail.gmail.com>
	<CAMjaK3A0aKF80nFCm-roRB1MC65RA3Rt-0QErx=6py8L4K7r_w@mail.gmail.com>
Message-ID: <CAA2gtHA5M1AqdsxSgwWQfXASymCS0qO6QEXXH6+aWddP2UgTOA@mail.gmail.com>

Could not be
because freeswitch is being run as root

And it can load /dev/ttyUSB0

sorry I had typo in audio line, It can load ttyUSB2 now





On Wed, Aug 22, 2012 at 6:03 PM, Dmitry Lysenko
<dvl36.ripe.nick at gmail.com>wrote:

> Maybe:
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/087009.html
> ?
>
> 2012/8/22 Sameer Khan <sameer2k3t at gmail.com>
>
>> this is wvdial output
>>
>>
>> Found a modem on /dev/ttyUSB0.
>> Modem configuration written to /etc/wvdial.conf.
>> ttyUSB0<Info>: Speed 9600; init "ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0"
>> ttyUSB3<Info>: Speed 9600; init "ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0"
>>
>>
>> On Wed, Aug 22, 2012 at 5:39 PM, Sameer Khan <sameer2k3t at gmail.com>wrote:
>>
>>> No luck
>>>
>>> 2012-08-22 17:35:38.337967 [ERR] gsmopen_protocol.cpp:3160 rev
>>> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
>>> ttyUSB2, NOT open
>>>
>>> 2012-08-22 17:37:27.057967 [ERR] gsmopen_protocol.cpp:3160 rev
>>> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
>>> ttyUSB1, NOT open
>>>
>>> Yes It is enabled, I tested before on other box
>>>
>>>
>>> On Wed, Aug 22, 2012 at 5:01 PM, Dmitry Lysenko <
>>> dvl36.ripe.nick at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> I have reproduced  such situation. It seems that
>>>> controldevice_audio_name is pointing to wrong /dev/ttyUSBx.
>>>> I suggest try:
>>>>
>>>>  <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>
>>>> if doesn't help try:
>>>>  <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>>>>
>>>> P.S. Are you sure modem voice feature enabled?
>>>>
>>>> 2012/8/22 Sameer Khan <sameer2k3t at gmail.com>
>>>>
>>>>> Hi,
>>>>>
>>>>> It says
>>>>>
>>>>> option                  8690  0
>>>>> usb_wwan                9960  1 option
>>>>> usbserial              31372  2 option,usb_wwan
>>>>>
>>>>> What should I do now to get it working ?
>>>>>
>>>>> Thanks for replying
>>>>>
>>>>> On Tue, Aug 21, 2012 at 2:55 PM, Dmitry Lysenko <
>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>
>>>>>> As root: lsmod | grep option
>>>>>> Should be something like that:
>>>>>> ---
>>>>>> root at debian:~# lsmod | grep option
>>>>>> option                 15351  6
>>>>>> usb_wwan                8362  1 option
>>>>>> usbserial              26535  14 usb_wwan,option
>>>>>> usbcore               122921  10
>>>>>> ehci_hcd,usb_storage,usbserial,gspca_main,usb_wwan,option,gspca_zc3xx,snd_usbmidi_lib,snd_usb_audio
>>>>>> ---
>>>>>>  2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>
>>>>>>> I guess so, I have CentOS 6 minimal installed
>>>>>>>
>>>>>>> how can I identify if this is option driver ?
>>>>>>>
>>>>>>> Yes I saw this message on ttyUSB0
>>>>>>>
>>>>>>> I can not load gsmopen with ttyUSB2 or ttyUSB1
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Aug 21, 2012 at 2:21 PM, Dmitry Lysenko <
>>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>>
>>>>>>>> Are you using "option" linux driver?
>>>>>>>> I checked Huawei E1550, E153 and E173 modems on some systems and
>>>>>>>> they have only 3 virtual serial ports.(using "option" driver)
>>>>>>>> Also you can check manually with minicom.
>>>>>>>> Usually ttyUSB0 is accepting  AT-commands, but will not reply with
>>>>>>>> OK or ERROR, ttyUSB1 is used for audio, and ttyUSB2 is accepting
>>>>>>>> AT-commands and replying  as normal serial modem.
>>>>>>>>
>>>>>>>>
>>>>>>>> 2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>>
>>>>>>>>> It didn't for me, according to wvdial my modem is ttyUSB0 and
>>>>>>>>> ttyUSB3
>>>>>>>>>
>>>>>>>>> On Mon, Aug 20, 2012 at 1:43 PM, Dmitry Lysenko <
>>>>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Hello.
>>>>>>>>>> Try:
>>>>>>>>>>  <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>>>>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>>>>>>> It's worked for me.
>>>>>>>>>>
>>>>>>>>>> 2012/8/20 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>>>>
>>>>>>>>>>>  Hi,
>>>>>>>>>>>
>>>>>>>>>>> I can't get audio working with e1550
>>>>>>>>>>>
>>>>>>>>>>> I have this config :
>>>>>>>>>>>     <interface id="1" name="gsm01">
>>>>>>>>>>>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>>>>>>>       <param name="controldevice_audio_name"
>>>>>>>>>>> value="/dev/ttyUSB3"/>
>>>>>>>>>>>
>>>>>>>>>>> I tried swaping 0 and 3 but having same results
>>>>>>>>>>>
>>>>>>>>>>> I am originating a call from softphone and answering it on cell
>>>>>>>>>>> phone but softphone doesn't get notified of being answered and keeps
>>>>>>>>>>> ringing
>>>>>>>>>>>
>>>>>>>>>>> Please help
>>>>>>>>>>>
>>>>>>>>>>> Thanks
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>>
>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sameer2k3t at gmail.com  Wed Aug 22 17:47:32 2012
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Wed, 22 Aug 2012 18:47:32 +0500
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAMjaK3BfjTzYkfdGo59MGo4fKxm=v0wVR6Ro_MKqGt-fcdf_+w@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
	<CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
	<CAA2gtHBXf-RfZ4eFB1k7NfCaYccqU6uK78NnxxBL0C=dO2PPBA@mail.gmail.com>
	<CAMjaK3DCzKEi7Gep9eZLhGs4FGix3edCDCxtymJXCwWHP9FSjA@mail.gmail.com>
	<CAA2gtHAcp_2+ch0-TOK-6XibCdmdFzfWOvJ9s2_7JKLjnf7MTA@mail.gmail.com>
	<CAMjaK3D+3XQDwU3s8m-FUj0wK_vPjOir3n3Lrvfwq9dS3Je2WQ@mail.gmail.com>
	<CAA2gtHBHkmUWM6EyqP14RtoH-jLUWitUdHXkZYKAVp3e54eBmw@mail.gmail.com>
	<CAMjaK3BfjTzYkfdGo59MGo4fKxm=v0wVR6Ro_MKqGt-fcdf_+w@mail.gmail.com>
Message-ID: <CAA2gtHDRE_JP3pe9bzr4_juh7=x_+U45sR4V4U9uooZnui+6ZA@mail.gmail.com>

I have the same too

2012-08-22 18:40:44.497064 [DEBUG] gsmopen_protocol.cpp:733 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  733  ][gsm01     ][-1, 0, 0]
Read line 0: |ERROR| la_counter=1
2012-08-22 18:40:44.497064 [DEBUG] gsmopen_protocol.cpp:1590 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1590 ][gsm01     ][-1, 0, 0]
got ERROR
2012-08-22 18:40:44.497064 [DEBUG] gsmopen_protocol.cpp:560 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  560  ][gsm01     ][-1, 0, 0]
at+sidet=0 does not get OK from the phone. Continuing.
2012-08-22 18:40:44.497064 [DEBUG] gsmopen_protocol.cpp:2165 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  2165 ][gsm01     ][-1, 0, 0]
sending: at+clvl=3, expecting: OK
2012-08-22 18:40:44.497064 [DEBUG] gsmopen_protocol.cpp:1918 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1918 ][gsm01     ][-1, 0, 0]
sent data... (a)
2012-08-22 18:40:44.517064 [DEBUG] gsmopen_protocol.cpp:1918 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1918 ][gsm01     ][-1, 0, 0]
sent data... (t)
2012-08-22 18:40:44.517064 [DEBUG] gsmopen_protocol.cpp:1918 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1918 ][gsm01     ][-1, 0, 0]
sent data... (+)
2012-08-22 18:40:44.517064 [DEBUG] gsmopen_protocol.cpp:1918 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1918 ][gsm01     ][-1, 0, 0]
sent data... (c)
2012-08-22 18:40:44.517064 [DEBUG] gsmopen_protocol.cpp:1918 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1918 ][gsm01     ][-1, 0, 0]
sent data... (l)
2012-08-22 18:40:44.517064 [DEBUG] gsmopen_protocol.cpp:1918 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1918 ][gsm01     ][-1, 0, 0]
sent data... (v)
2012-08-22 18:40:44.517064 [DEBUG] gsmopen_protocol.cpp:1918 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1918 ][gsm01     ][-1, 0, 0]
sent data... (l)
2012-08-22 18:40:44.517064 [DEBUG] gsmopen_protocol.cpp:1918 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1918 ][gsm01     ][-1, 0, 0]
sent data... (=)
2012-08-22 18:40:44.517064 [DEBUG] gsmopen_protocol.cpp:1918 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1918 ][gsm01     ][-1, 0, 0]
sent data... (3)
2012-08-22 18:40:44.517064 [DEBUG] gsmopen_protocol.cpp:1966 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1966 ][gsm01     ][-1, 0, 0]
sent (carriage return)
2012-08-22 18:40:44.537064 [DEBUG] gsmopen_protocol.cpp:733 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  733  ][gsm01     ][-1, 0, 0]
Read line 0: |OK| la_counter=1
2012-08-22 18:40:44.537064 [DEBUG] gsmopen_protocol.cpp:1578 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1578 ][gsm01     ][-1, 0, 0]
|OK| got what EXPECTED
2012-08-22 18:40:44.537064 [DEBUG] gsmopen_protocol.cpp:3158 rev
ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  3158 ][gsm01     ][-1, 0, 0]
port /dev/ttyUSB2, SUCCESS open


but when I call to the sim, freeswitch doesn't accepts call, I can't see
any call coming on cli

I will now originate a call from this sim and will see if I get audio

On Wed, Aug 22, 2012 at 6:35 PM, Dmitry Lysenko
<dvl36.ripe.nick at gmail.com>wrote:

>
>
> 2012/8/22 Sameer Khan <sameer2k3t at gmail.com>
>
>> No luck
>>
>> 2012-08-22 17:35:38.337967 [ERR] gsmopen_protocol.cpp:3160 rev
>> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
>> ttyUSB2, NOT open
>>
>
> Sameer, should be */dev/*ttyUSB2
>
>
> From Mi Ke log:
>
> 2012-08-21 09:18:18.973501 [ERR] gsmopen_protocol.cpp:122 rev ccae5cd|07bc7ba[(nil)|37 ][ERRORA 122 ][gsm00 ][-1, 0, 0] port */dev/ttyUSB0*, NOT open
>
> And my log:
>
> gsmopen_protocol.cpp:120 rev ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  120  ][gsm01     ][-1, 0, 0] port */dev/ttyUSB2*, SUCCESS open
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lconroy at insensate.co.uk  Wed Aug 22 16:54:40 2012
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Wed, 22 Aug 2012 13:54:40 +0100
Subject: [Freeswitch-users] Build on Apple PPC?
Message-ID: <6B206226-751D-43F2-B214-614A0009BF38@insensate.co.uk>

Hi there,
I'm trying to build 1.21 stable from git (replacing my current 1.06 tarball build) on a TiBook (PPC G4) running Leopard (10.5.8).

It's failing during configure -- "configure: error: no usable libodbc; please install unixodbc devel package or equivalent".
I used the command line flag --disable-core-odbc-support, so don't understand ... why is configure looking for odbc at all?

anyone have an idea why this flag is ignored?

all the best,
  Lawrence

From ananiev at svyaz.com  Wed Aug 22 17:37:18 2012
From: ananiev at svyaz.com (Vladimir Ananiev)
Date: Wed, 22 Aug 2012 17:37:18 +0400
Subject: [Freeswitch-users] Wrong transport=tcp in SIP Contact field.
Message-ID: <CE660A7C-9B8A-4AD8-B499-41E402D674B9@svyaz.com>

Hello all.

I have a problem with some specific softphone connecting to the Freswitch system.
This softphone connects with UDP protocol, but sends additional ;transport=tcp line in the end of Contact field, that makes freeswitch thinking that the transport is really TCP, not the UDP. And after that incoming calls are not working. Outgoing calls from that softphone are working just fine.

Is there a way to force freeswitch to use UDP in this case ? I already have <param name="bind-params" value="transport=udp" /> in sip profile settings...

See the packet dump from wireshark:
User Datagram Protocol, Src Port: na-localise (5062), Dst Port: sip (5060)
    Source port: na-localise (5062)
    Destination port: sip (5060)
    Length: 793
    Checksum: 0x34ff [validation disabled]
Session Initiation Protocol
    Request-Line: REGISTER sip:sip.test.com SIP/2.0
        Method: REGISTER
        Request-URI: sip:sip.test.com
        [Resent Packet: False]
    Message Header
        CSeq: 535 REGISTER
        Via: SIP/2.0/UDP 192.168.0.12:5062;branch=z9hG4bK2edb4d2b-d007-1910-9210-1cc1de60bc3c;rport
        User-Agent: TMSoftPhone
        [truncated] Authorization: Digest username="621", realm="sip.test.com", nonce="c4f876ae-eb58-11e1-8919-6df0d0a4f1d4", uri="sip:sip.test.com", algorithm=MD5, response="52625726279aaf004b61871164b61088", cnonce="0560cd01-d007-1910-91f
        From: <sip:621 at sip.test.com>;tag=25ffcc01-d007-1910-91fb-1cc1de60bc3c
        Call-ID: 25ffcc01-d007-1910-91fa-1cc1de60bc3c
        To: <sip:621 at sip.test.com>
        Contact: <sip:621 at 192.168.0.12:5062;transport=tcp>
        Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
        Expires: 120
        Content-Length: 0
        Max-Forwards: 70

---
??????? ???????? ??????????
??????????? ???????? 
??? "?????"
???.   +7 495 7238000, ???. 101
????.+7 495 7308111
E-mail: ananiev at svyaz.com



From dvl36.ripe.nick at gmail.com  Wed Aug 22 17:49:36 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Wed, 22 Aug 2012 16:49:36 +0300
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAA2gtHA5M1AqdsxSgwWQfXASymCS0qO6QEXXH6+aWddP2UgTOA@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
	<CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
	<CAA2gtHBXf-RfZ4eFB1k7NfCaYccqU6uK78NnxxBL0C=dO2PPBA@mail.gmail.com>
	<CAMjaK3DCzKEi7Gep9eZLhGs4FGix3edCDCxtymJXCwWHP9FSjA@mail.gmail.com>
	<CAA2gtHAcp_2+ch0-TOK-6XibCdmdFzfWOvJ9s2_7JKLjnf7MTA@mail.gmail.com>
	<CAMjaK3D+3XQDwU3s8m-FUj0wK_vPjOir3n3Lrvfwq9dS3Je2WQ@mail.gmail.com>
	<CAA2gtHBHkmUWM6EyqP14RtoH-jLUWitUdHXkZYKAVp3e54eBmw@mail.gmail.com>
	<CAA2gtHCpqRFB+d8xE5DWYbuC2U8uaWV_56VUzpSh7t2zzUvtHQ@mail.gmail.com>
	<CAMjaK3A0aKF80nFCm-roRB1MC65RA3Rt-0QErx=6py8L4K7r_w@mail.gmail.com>
	<CAA2gtHA5M1AqdsxSgwWQfXASymCS0qO6QEXXH6+aWddP2UgTOA@mail.gmail.com>
Message-ID: <CAMjaK3BFRXcKrBXHz2Z7zxOTm-AW2XfkN2BBXDNLFhQgQ1Csvw@mail.gmail.com>

So, if it doesn't work with audio device at /dev/ttyUSB2, try /dev/ttyUSB1
Typo was at ttyUSB1 too.

2012/8/22 Sameer Khan <sameer2k3t at gmail.com>

> Could not be
> because freeswitch is being run as root
>
> And it can load /dev/ttyUSB0
>
> sorry I had typo in audio line, It can load ttyUSB2 now
>
>
>
>
>
> On Wed, Aug 22, 2012 at 6:03 PM, Dmitry Lysenko <dvl36.ripe.nick at gmail.com
> > wrote:
>
>> Maybe:
>>
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/087009.html
>> ?
>>
>> 2012/8/22 Sameer Khan <sameer2k3t at gmail.com>
>>
>>> this is wvdial output
>>>
>>>
>>> Found a modem on /dev/ttyUSB0.
>>> Modem configuration written to /etc/wvdial.conf.
>>> ttyUSB0<Info>: Speed 9600; init "ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0"
>>> ttyUSB3<Info>: Speed 9600; init "ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0"
>>>
>>>
>>> On Wed, Aug 22, 2012 at 5:39 PM, Sameer Khan <sameer2k3t at gmail.com>wrote:
>>>
>>>> No luck
>>>>
>>>> 2012-08-22 17:35:38.337967 [ERR] gsmopen_protocol.cpp:3160 rev
>>>> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
>>>> ttyUSB2, NOT open
>>>>
>>>> 2012-08-22 17:37:27.057967 [ERR] gsmopen_protocol.cpp:3160 rev
>>>> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
>>>> ttyUSB1, NOT open
>>>>
>>>> Yes It is enabled, I tested before on other box
>>>>
>>>>
>>>> On Wed, Aug 22, 2012 at 5:01 PM, Dmitry Lysenko <
>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> I have reproduced  such situation. It seems that
>>>>> controldevice_audio_name is pointing to wrong /dev/ttyUSBx.
>>>>> I suggest try:
>>>>>
>>>>>  <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>>
>>>>> if doesn't help try:
>>>>>  <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>>>>>
>>>>> P.S. Are you sure modem voice feature enabled?
>>>>>
>>>>> 2012/8/22 Sameer Khan <sameer2k3t at gmail.com>
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> It says
>>>>>>
>>>>>> option                  8690  0
>>>>>> usb_wwan                9960  1 option
>>>>>> usbserial              31372  2 option,usb_wwan
>>>>>>
>>>>>> What should I do now to get it working ?
>>>>>>
>>>>>> Thanks for replying
>>>>>>
>>>>>> On Tue, Aug 21, 2012 at 2:55 PM, Dmitry Lysenko <
>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>
>>>>>>> As root: lsmod | grep option
>>>>>>> Should be something like that:
>>>>>>> ---
>>>>>>> root at debian:~# lsmod | grep option
>>>>>>> option                 15351  6
>>>>>>> usb_wwan                8362  1 option
>>>>>>> usbserial              26535  14 usb_wwan,option
>>>>>>> usbcore               122921  10
>>>>>>> ehci_hcd,usb_storage,usbserial,gspca_main,usb_wwan,option,gspca_zc3xx,snd_usbmidi_lib,snd_usb_audio
>>>>>>> ---
>>>>>>>  2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>
>>>>>>>> I guess so, I have CentOS 6 minimal installed
>>>>>>>>
>>>>>>>> how can I identify if this is option driver ?
>>>>>>>>
>>>>>>>> Yes I saw this message on ttyUSB0
>>>>>>>>
>>>>>>>> I can not load gsmopen with ttyUSB2 or ttyUSB1
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Aug 21, 2012 at 2:21 PM, Dmitry Lysenko <
>>>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Are you using "option" linux driver?
>>>>>>>>> I checked Huawei E1550, E153 and E173 modems on some systems and
>>>>>>>>> they have only 3 virtual serial ports.(using "option" driver)
>>>>>>>>> Also you can check manually with minicom.
>>>>>>>>> Usually ttyUSB0 is accepting  AT-commands, but will not reply with
>>>>>>>>> OK or ERROR, ttyUSB1 is used for audio, and ttyUSB2 is accepting
>>>>>>>>> AT-commands and replying  as normal serial modem.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> 2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>>>
>>>>>>>>>> It didn't for me, according to wvdial my modem is ttyUSB0 and
>>>>>>>>>> ttyUSB3
>>>>>>>>>>
>>>>>>>>>> On Mon, Aug 20, 2012 at 1:43 PM, Dmitry Lysenko <
>>>>>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Hello.
>>>>>>>>>>> Try:
>>>>>>>>>>>  <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>>>>>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>>>>>>>> It's worked for me.
>>>>>>>>>>>
>>>>>>>>>>> 2012/8/20 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>>>>>
>>>>>>>>>>>>  Hi,
>>>>>>>>>>>>
>>>>>>>>>>>> I can't get audio working with e1550
>>>>>>>>>>>>
>>>>>>>>>>>> I have this config :
>>>>>>>>>>>>     <interface id="1" name="gsm01">
>>>>>>>>>>>>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>>>>>>>>       <param name="controldevice_audio_name"
>>>>>>>>>>>> value="/dev/ttyUSB3"/>
>>>>>>>>>>>>
>>>>>>>>>>>> I tried swaping 0 and 3 but having same results
>>>>>>>>>>>>
>>>>>>>>>>>> I am originating a call from softphone and answering it on cell
>>>>>>>>>>>> phone but softphone doesn't get notified of being answered and keeps
>>>>>>>>>>>> ringing
>>>>>>>>>>>>
>>>>>>>>>>>> Please help
>>>>>>>>>>>>
>>>>>>>>>>>> Thanks
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>>
>>>>>>>>>>>> 
>>>>>>>>>>>> 
>>>>>>>>>>>>
>>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>>
>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at endigotech.com  Wed Aug 22 18:05:26 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 22 Aug 2012 10:05:26 -0400
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <CD9593DE-1F02-4135-AEDD-7ED7C6A5FB22@visionutveckling.se>
References: <5034019b.c456b60a.3fd2.ffffdc5cSMTPIN_ADDED@mx.google.com>
	<CALNZuwmN+xfGkW0HqqDpXdJ4WaW5o59BcX-RiuKdNo18QH5D7g@mail.gmail.com>
	<CD9593DE-1F02-4135-AEDD-7ED7C6A5FB22@visionutveckling.se>
Message-ID: <CAKV-jFxPPfRwBcZZDgxz2uj79OSbNW=SHRKLz_kM42CMD97WaA@mail.gmail.com>

Hardware is cheap.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 22, 2012 2:14 AM, "Peter Olsson" <peter.olsson at visionutveckling.se>
wrote:

> If your current hardware doesn't support more RAM - buy a new server. FS
> scales well, but it will need the memory and CPU to perform.
>
> To redesign FS to not create a new thread per channel would be a major
> rewrite of the entire core.
>
> Considering your requirements it sonds like Kamailio or OpenSIPS is a
> better alternative for you, especially since you don't want to hande media
> anyway.
>
> /Peter
>
> 22 aug 2012 kl. 00:39 skrev "Jason Caulfield" <
> jason.caulfield at intermetro.net<mailto:jason.caulfield at intermetro.net>>:
>
> It is not so much the cps but the concurrent calls.
>
> I am looking at maintaining a 30 cps with a 5 min. avg. call length which
> results in a sustained 9000 concurrent calls.
>
> This results in 85% mem. usage on a 4 GB machine, However CPU is at 25%
> (total) on a dual core with hyper threading.
>
> Current hardware does not support increase memory.
>
> I am striving for about 75% cpu (appox. 90 cps).
>
> Jason
>
>
> From: 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 Avi Marcus
> Sent: Tuesday, August 21, 2012 2:58 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FreeSWITCH Question
>
> Do you have such a large cps/concurrent volume that you're actually seeing
> a performance hit?
>
> -Avi
>
> On Wed, Aug 22, 2012 at 12:35 AM, Jason Caulfield <
> jason.caulfield at intermetro.net<mailto:jason.caulfield at intermetro.net>>
> wrote:
> I am using FreeSWITCH in media bypass mode to decrease memory and cpu
> usage.
>
> I notice that mod_sofia creates threads for each call leg to maintain
> session context.
>
> Do you know of a way to configure FreeSWITCH to use a table to maintain
> the context when in media bypass mode to reduce the number of threads?
>
> I am hoping that this will speed things up by reducing thread context
> switching and reduce memory usage by decreasing memory allocation for each
> thread.
>
> Thanks for the help,
> Jason
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> 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:50340a7d32766392456629!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> 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:50340a7d32766392456629!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lists at telefaks.de  Wed Aug 22 18:07:48 2012
From: lists at telefaks.de (Peter Steinbach)
Date: Wed, 22 Aug 2012 16:07:48 +0200
Subject: [Freeswitch-users] displace_session and moh via event socket
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD073ECD1637@NY1-EXMB-01.ip-soft.net>
References: <502E5D31.9010203@telefaks.de>
	<6A6B4C284AD15042B429EB9D904544AD073ECD1637@NY1-EXMB-01.ip-soft.net>
Message-ID: <5034E7B4.1010406@telefaks.de>

Hello Hector,

thanks for your hint. This worked. Before playing the voice files I now
sent a

sched_api +10 uuid_break <uuid>

via event_socket.
Now I have exactly the desired behaviour.

Best regards
Peter


On 08/17/12 18:34, Hector Geraldino wrote:
> Have you tried a combination of playback + sched_api +10 uuid break ?
>
> IIRC you can use break app to cancel the playback, so all you need is to schedule it to happen sometime in the future (10 seconds in your case)
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Steinbach
> Sent: Friday, August 17, 2012 11:03 AM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] displace_session and moh via event socket
>
> Hello,
>
> In an IVR we want to play moh for 10 seconds via event socket to an existing session. We send the following message (we actually do not use ESL for certain reasons):
>   SendMsg
>   call-command: execute
>   execute-app-name: displace_session
>   execute-app-arg: local_stream://moh m +10000
>   event-lock:false
>
> I think, this is the only way to play a stream for a certain time within a session right?
>
> We get the following output:
> 2012-08-17 16:05:20.261460 [DEBUG] switch_ivr.c:599 sofia/internal/202 at fs100.mydomain.de Command Execute displace_session(local_stream://moh m +10000) EXECUTE sofia/internal/202 at fs100.mydomain.de
> displace_session(local_stream://moh m +10000)
> 2012-08-17 16:05:20.261460 [DEBUG] mod_local_stream.c:417 Opening Stream [moh/8000] 8000hz
> 2012-08-17 16:05:20.261460 [DEBUG] switch_core_media_bug.c:502 Attaching BUG to sofia/internal/202 at fs100.mydomain.de
> 2012-08-17 16:05:20.321460 [DEBUG] switch_core_io.c:353 Setting BUG Codec PCMA:8
> 2012-08-17 16:05:27.981463 [CRIT] mod_local_stream.c:297 Leaking stream handle! [switch_ivr_displace_session() src/switch_ivr_async.c:951]
> 2012-08-17 16:05:35.701462 [CRIT] mod_local_stream.c:297 Leaking stream handle! [switch_ivr_displace_session() src/switch_ivr_async.c:951]
> 2012-08-17 16:05:43.421463 [CRIT] mod_local_stream.c:297 Leaking stream handle! [switch_ivr_displace_session() src/switch_ivr_async.c:951] and no sound is played.
>
> Passing a regular filename does not work either. Without the "m" or time parameter we have also the same result: No file is played. See the following log:
> 2012-08-17 17:00:51.911469 [DEBUG] switch_ivr.c:599 sofia/internal/202 at fs100.mydomain.de Command Execute
> displace_session(/usr/local/freeswitch/sounds/en/us/callie/google/de/willkommen.mp3
> m +10000)
> EXECUTE sofia/internal/202 at fs100.mydomain.de
> displace_session(/usr/local/freeswitch/sounds/en/us/callie/google/de/willkommen.mp3
> m +10000)
> 2012-08-17 17:00:51.921461 [DEBUG] switch_core_media_bug.c:502 Attaching BUG to sofia/internal/202 at fs100.mydomain.de
> 2012-08-17 17:00:51.941460 [DEBUG] switch_core_io.c:353 Setting BUG Codec PCMA:8 And then output stops for this session.
>
> An normal playback works however, but with playback we cannot limit a file to a certain time period, right?
>
> Did Anybody have the same issue and knows how to solve this?
>
>
> --
> With kind regards
> Peter Steinbach 
>
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
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From lists at kavun.ch  Wed Aug 22 18:23:55 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 22 Aug 2012 10:23:55 -0400
Subject: [Freeswitch-users] Concatenated multiple playback types
Message-ID: <ABE29461-13EE-431E-ADCC-12E4BA7362FC@kavun.ch>

Hello there,

I am trying to add some silence before the alone sound file in my conference.cnf.xml and can't get it to work.

Here is what I tried:
<param name="alone-sound" value="silence_stream://5000!file_string://conference/conf-alone.wav"/>
<param name="alone-sound" value="silence_stream://5000&file_string://conference/conf-alone.wav"/>
<param name="alone-sound" value="silence_stream://5000|file_string://conference/conf-alone.wav"/>
<param name="alone-sound" value="silence_stream://5000;file_string://conference/conf-alone.wav"/>

Any idea on how to do this?

Thanks!
Emrah

From sameer2k3t at gmail.com  Wed Aug 22 18:32:08 2012
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Wed, 22 Aug 2012 19:32:08 +0500
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAMjaK3BFRXcKrBXHz2Z7zxOTm-AW2XfkN2BBXDNLFhQgQ1Csvw@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
	<CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
	<CAA2gtHBXf-RfZ4eFB1k7NfCaYccqU6uK78NnxxBL0C=dO2PPBA@mail.gmail.com>
	<CAMjaK3DCzKEi7Gep9eZLhGs4FGix3edCDCxtymJXCwWHP9FSjA@mail.gmail.com>
	<CAA2gtHAcp_2+ch0-TOK-6XibCdmdFzfWOvJ9s2_7JKLjnf7MTA@mail.gmail.com>
	<CAMjaK3D+3XQDwU3s8m-FUj0wK_vPjOir3n3Lrvfwq9dS3Je2WQ@mail.gmail.com>
	<CAA2gtHBHkmUWM6EyqP14RtoH-jLUWitUdHXkZYKAVp3e54eBmw@mail.gmail.com>
	<CAA2gtHCpqRFB+d8xE5DWYbuC2U8uaWV_56VUzpSh7t2zzUvtHQ@mail.gmail.com>
	<CAMjaK3A0aKF80nFCm-roRB1MC65RA3Rt-0QErx=6py8L4K7r_w@mail.gmail.com>
	<CAA2gtHA5M1AqdsxSgwWQfXASymCS0qO6QEXXH6+aWddP2UgTOA@mail.gmail.com>
	<CAMjaK3BFRXcKrBXHz2Z7zxOTm-AW2XfkN2BBXDNLFhQgQ1Csvw@mail.gmail.com>
Message-ID: <CAA2gtHAfQBGPeUxt+Ckko84+NCwOyD=wocFvOUgA+Op7hUeu=w@mail.gmail.com>

It worked with

    <param name="controldevice_name" value="/dev/ttyUSB3"/>
      <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>


I tried usermod -a -G dialout root

before typo as freeswitch was running with root, so may it fixed the issue.

Thanks alot for your help my friend



On Wed, Aug 22, 2012 at 6:49 PM, Dmitry Lysenko
<dvl36.ripe.nick at gmail.com>wrote:

> So, if it doesn't work with audio device at /dev/ttyUSB2, try /dev/ttyUSB1
> Typo was at ttyUSB1 too.
>
> 2012/8/22 Sameer Khan <sameer2k3t at gmail.com>
>
>> Could not be
>> because freeswitch is being run as root
>>
>> And it can load /dev/ttyUSB0
>>
>> sorry I had typo in audio line, It can load ttyUSB2 now
>>
>>
>>
>>
>>
>> On Wed, Aug 22, 2012 at 6:03 PM, Dmitry Lysenko <
>> dvl36.ripe.nick at gmail.com> wrote:
>>
>>> Maybe:
>>>
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/087009.html
>>> ?
>>>
>>> 2012/8/22 Sameer Khan <sameer2k3t at gmail.com>
>>>
>>>> this is wvdial output
>>>>
>>>>
>>>> Found a modem on /dev/ttyUSB0.
>>>> Modem configuration written to /etc/wvdial.conf.
>>>> ttyUSB0<Info>: Speed 9600; init "ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0"
>>>> ttyUSB3<Info>: Speed 9600; init "ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0"
>>>>
>>>>
>>>> On Wed, Aug 22, 2012 at 5:39 PM, Sameer Khan <sameer2k3t at gmail.com>wrote:
>>>>
>>>>> No luck
>>>>>
>>>>> 2012-08-22 17:35:38.337967 [ERR] gsmopen_protocol.cpp:3160 rev
>>>>> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
>>>>> ttyUSB2, NOT open
>>>>>
>>>>> 2012-08-22 17:37:27.057967 [ERR] gsmopen_protocol.cpp:3160 rev
>>>>> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
>>>>> ttyUSB1, NOT open
>>>>>
>>>>> Yes It is enabled, I tested before on other box
>>>>>
>>>>>
>>>>> On Wed, Aug 22, 2012 at 5:01 PM, Dmitry Lysenko <
>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I have reproduced  such situation. It seems that
>>>>>> controldevice_audio_name is pointing to wrong /dev/ttyUSBx.
>>>>>> I suggest try:
>>>>>>
>>>>>>  <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>>>
>>>>>> if doesn't help try:
>>>>>>  <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>>>>>>
>>>>>> P.S. Are you sure modem voice feature enabled?
>>>>>>
>>>>>> 2012/8/22 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> It says
>>>>>>>
>>>>>>> option                  8690  0
>>>>>>> usb_wwan                9960  1 option
>>>>>>> usbserial              31372  2 option,usb_wwan
>>>>>>>
>>>>>>> What should I do now to get it working ?
>>>>>>>
>>>>>>> Thanks for replying
>>>>>>>
>>>>>>> On Tue, Aug 21, 2012 at 2:55 PM, Dmitry Lysenko <
>>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>>
>>>>>>>> As root: lsmod | grep option
>>>>>>>> Should be something like that:
>>>>>>>> ---
>>>>>>>> root at debian:~# lsmod | grep option
>>>>>>>> option                 15351  6
>>>>>>>> usb_wwan                8362  1 option
>>>>>>>> usbserial              26535  14 usb_wwan,option
>>>>>>>> usbcore               122921  10
>>>>>>>> ehci_hcd,usb_storage,usbserial,gspca_main,usb_wwan,option,gspca_zc3xx,snd_usbmidi_lib,snd_usb_audio
>>>>>>>> ---
>>>>>>>>  2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>>
>>>>>>>>> I guess so, I have CentOS 6 minimal installed
>>>>>>>>>
>>>>>>>>> how can I identify if this is option driver ?
>>>>>>>>>
>>>>>>>>> Yes I saw this message on ttyUSB0
>>>>>>>>>
>>>>>>>>> I can not load gsmopen with ttyUSB2 or ttyUSB1
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Tue, Aug 21, 2012 at 2:21 PM, Dmitry Lysenko <
>>>>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Are you using "option" linux driver?
>>>>>>>>>> I checked Huawei E1550, E153 and E173 modems on some systems and
>>>>>>>>>> they have only 3 virtual serial ports.(using "option" driver)
>>>>>>>>>> Also you can check manually with minicom.
>>>>>>>>>> Usually ttyUSB0 is accepting  AT-commands, but will not reply
>>>>>>>>>> with OK or ERROR, ttyUSB1 is used for audio, and ttyUSB2 is accepting
>>>>>>>>>> AT-commands and replying  as normal serial modem.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> 2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>>>>
>>>>>>>>>>> It didn't for me, according to wvdial my modem is ttyUSB0 and
>>>>>>>>>>> ttyUSB3
>>>>>>>>>>>
>>>>>>>>>>> On Mon, Aug 20, 2012 at 1:43 PM, Dmitry Lysenko <
>>>>>>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> Hello.
>>>>>>>>>>>> Try:
>>>>>>>>>>>>  <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>>>>>>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>>>>>>>>> It's worked for me.
>>>>>>>>>>>>
>>>>>>>>>>>> 2012/8/20 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>>>>>>
>>>>>>>>>>>>>  Hi,
>>>>>>>>>>>>>
>>>>>>>>>>>>> I can't get audio working with e1550
>>>>>>>>>>>>>
>>>>>>>>>>>>> I have this config :
>>>>>>>>>>>>>     <interface id="1" name="gsm01">
>>>>>>>>>>>>>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>>>>>>>>>       <param name="controldevice_audio_name"
>>>>>>>>>>>>> value="/dev/ttyUSB3"/>
>>>>>>>>>>>>>
>>>>>>>>>>>>> I tried swaping 0 and 3 but having same results
>>>>>>>>>>>>>
>>>>>>>>>>>>> I am originating a call from softphone and answering it on
>>>>>>>>>>>>> cell phone but softphone doesn't get notified of being answered and keeps
>>>>>>>>>>>>> ringing
>>>>>>>>>>>>>
>>>>>>>>>>>>> Please help
>>>>>>>>>>>>>
>>>>>>>>>>>>> Thanks
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>>>
>>>>>>>>>>>>> 
>>>>>>>>>>>>> 
>>>>>>>>>>>>>
>>>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>>
>>>>>>>>>>>> 
>>>>>>>>>>>> 
>>>>>>>>>>>>
>>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>>
>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 krice at freeswitch.org  Wed Aug 22 18:33:11 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 22 Aug 2012 09:33:11 -0500
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <35C0D459-4C78-476F-B516-56BD8687DE54@digitalmail.com>
Message-ID: <CC5A57D7.3C60A%krice@freeswitch.org>

90cps on a 4G ram machine isnt that bad... Its the 9000 concurrent calls
that is the real question here...

K


On 8/22/12 12:47 AM, "Alex Lake" <alex at thewinelake.com> wrote:

> You want to do 90cps on a machine that has just 4G RAM?!
> 
> 
> 
> On 21 Aug 2012, at 23:38, "Jason Caulfield" <jason.caulfield at intermetro.net>
> wrote:
> 
>> It is not so much the cps but the concurrent calls.
>>  
>> I am looking at maintaining a 30 cps with a 5 min. avg. call length which
>> results in a sustained 9000 concurrent calls.
>>  
>> This results in 85% mem. usage on a 4 GB machine, However CPU is at 25%
>> (total) on a dual core with hyper threading.
>>  
>> Current hardware does not support increase memory.
>>  
>> I am striving for about 75% cpu (appox. 90 cps).
>>  
>> Jason
>>  
>>  
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
>> Marcus
>> Sent: Tuesday, August 21, 2012 2:58 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] FreeSWITCH Question
>>  
>> 
>> Do you have such a large cps/concurrent volume that you're actually seeing a
>> performance hit?
>> 
>> 
>> -Avi
>>  
>> 
>> On Wed, Aug 22, 2012 at 12:35 AM, Jason Caulfield
>> <jason.caulfield at intermetro.net> wrote:
>> 
>> I am using FreeSWITCH in media bypass mode to decrease memory and cpu usage.
>>  
>> I notice that mod_sofia creates threads for each call leg to maintain session
>> context.
>>  
>> Do you know of a way to configure FreeSWITCH to use a table to maintain the
>> context when in media bypass mode to reduce the number of threads?
>>  
>> I am hoping that this will speed things up by reducing thread context
>> switching and reduce memory usage by decreasing memory allocation for each
>> thread.
>>  
>> Thanks for the help,
>> Jason
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>  
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From peter.olsson at visionutveckling.se  Wed Aug 22 18:36:17 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 22 Aug 2012 14:36:17 +0000
Subject: [Freeswitch-users] Concatenated multiple playback types
Message-ID: <1FFF97C269757C458224B7C895F35F1514B3F0@cantor.std.visionutv.se>

I don't believe this is possible. If you record 5 seconds of silence into a file you might be able to use file_string:// to play them both.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Emrah
Skickat: den 22 augusti 2012 16:24
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Concatenated multiple playback types

Hello there,

I am trying to add some silence before the alone sound file in my conference.cnf.xml and can't get it to work.

Here is what I tried:
<param name="alone-sound" value="silence_stream://5000!file_string://conference/conf-alone.wav"/>
<param name="alone-sound" value="silence_stream://5000&file_string://conference/conf-alone.wav"/>
<param name="alone-sound" value="silence_stream://5000|file_string://conference/conf-alone.wav"/>
<param name="alone-sound" value="silence_stream://5000;file_string://conference/conf-alone.wav"/>

Any idea on how to do this?

Thanks!
Emrah
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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!DSPAM:5034e9a632761303016055!



From anthony.minessale at gmail.com  Wed Aug 22 18:43:15 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 22 Aug 2012 09:43:15 -0500
Subject: [Freeswitch-users] Wrong transport=tcp in SIP Contact field.
In-Reply-To: <CE660A7C-9B8A-4AD8-B499-41E402D674B9@svyaz.com>
References: <CE660A7C-9B8A-4AD8-B499-41E402D674B9@svyaz.com>
Message-ID: <CAKbxfG91+q+qUGkRgrJTJV=CeoVkHaE55x31BK1qHKmRUv32aA@mail.gmail.com>

You could use iptables or a sip proxy to rewrite it.


On Wed, Aug 22, 2012 at 8:37 AM, Vladimir Ananiev <ananiev at svyaz.com> wrote:
> Hello all.
>
> I have a problem with some specific softphone connecting to the Freswitch system.
> This softphone connects with UDP protocol, but sends additional ;transport=tcp line in the end of Contact field, that makes freeswitch thinking that the transport is really TCP, not the UDP. And after that incoming calls are not working. Outgoing calls from that softphone are working just fine.
>
> Is there a way to force freeswitch to use UDP in this case ? I already have <param name="bind-params" value="transport=udp" /> in sip profile settings...
>
> See the packet dump from wireshark:
> User Datagram Protocol, Src Port: na-localise (5062), Dst Port: sip (5060)
>     Source port: na-localise (5062)
>     Destination port: sip (5060)
>     Length: 793
>     Checksum: 0x34ff [validation disabled]
> Session Initiation Protocol
>     Request-Line: REGISTER sip:sip.test.com SIP/2.0
>         Method: REGISTER
>         Request-URI: sip:sip.test.com
>         [Resent Packet: False]
>     Message Header
>         CSeq: 535 REGISTER
>         Via: SIP/2.0/UDP 192.168.0.12:5062;branch=z9hG4bK2edb4d2b-d007-1910-9210-1cc1de60bc3c;rport
>         User-Agent: TMSoftPhone
>         [truncated] Authorization: Digest username="621", realm="sip.test.com", nonce="c4f876ae-eb58-11e1-8919-6df0d0a4f1d4", uri="sip:sip.test.com", algorithm=MD5, response="52625726279aaf004b61871164b61088", cnonce="0560cd01-d007-1910-91f
>         From: <sip:621 at sip.test.com>;tag=25ffcc01-d007-1910-91fb-1cc1de60bc3c
>         Call-ID: 25ffcc01-d007-1910-91fa-1cc1de60bc3c
>         To: <sip:621 at sip.test.com>
>         Contact: <sip:621 at 192.168.0.12:5062;transport=tcp>
>         Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
>         Expires: 120
>         Content-Length: 0
>         Max-Forwards: 70
>
> ---
> ??????? ???????? ??????????
> ??????????? ????????
> ??? "?????"
> ???.   +7 495 7238000, ???. 101
> ????.+7 495 7308111
> E-mail: ananiev at svyaz.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

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

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

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


From ihalliday at ndevix.com  Wed Aug 22 18:52:17 2012
From: ihalliday at ndevix.com (Ian Halliday)
Date: Wed, 22 Aug 2012 09:52:17 -0500
Subject: [Freeswitch-users] Routing inbound calls to extentions with did
	intact
Message-ID: <6293150F220E44F5A0438104F2A46271@NDX0>

Hello All,

I use freeswitch to route inbound calls to some of the PBX's we host, and I
use the following method (Below) to do so. It's been working great (20,000
calls in the first month without a problem), but I wanted to get a second
opinion to see if there was a better way to do such a thing (keeping the DID
intact). Any help would be appreciated. (I used an example DID and extention)

<extension name="setdid">
<condition field="destination_number" expression="^setdid$">
<action application="set" data="dialed_ext=${did}"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="disable_q850_reason=true"/>
<action application="bridge"
data="sofia/internal/${did}${regex(${sofia_contact(${user})}|^[^\@]+(.*)|%1)}"/>
</condition>
</extension>

<extension name="13125551234">
<condition field="destination_number" expression="^13125551234$">
<action application="set" data="user=user1002"/>
<action application="set" data="did=${destination_number}"/>
<action application="execute_extension" data="setdid"/>
</condition>
</extension>

Thanks!
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From ananiev at svyaz.com  Wed Aug 22 18:52:51 2012
From: ananiev at svyaz.com (Vladimir Ananiev)
Date: Wed, 22 Aug 2012 18:52:51 +0400
Subject: [Freeswitch-users] Wrong transport=tcp in SIP Contact field.
In-Reply-To: <CAKbxfG91+q+qUGkRgrJTJV=CeoVkHaE55x31BK1qHKmRUv32aA@mail.gmail.com>
References: <CE660A7C-9B8A-4AD8-B499-41E402D674B9@svyaz.com>
	<CAKbxfG91+q+qUGkRgrJTJV=CeoVkHaE55x31BK1qHKmRUv32aA@mail.gmail.com>
Message-ID: <2C6886BF-76C0-4065-A42B-21014FB2630C@svyaz.com>

Thanks for the idea, I will try to use iptables for this.

22.08.2012, ? 18:43, Anthony Minessale <anthony.minessale at gmail.com> ???????(?):

> You could use iptables or a sip proxy to rewrite it.
> 
> 
> On Wed, Aug 22, 2012 at 8:37 AM, Vladimir Ananiev <ananiev at svyaz.com> wrote:
>> Hello all.
>> 
>> I have a problem with some specific softphone connecting to the Freswitch system.
>> This softphone connects with UDP protocol, but sends additional ;transport=tcp line in the end of Contact field, that makes freeswitch thinking that the transport is really TCP, not the UDP. And after that incoming calls are not working. Outgoing calls from that softphone are working just fine.
>> 
>> Is there a way to force freeswitch to use UDP in this case ? I already have <param name="bind-params" value="transport=udp" /> in sip profile settings...
>> 
>> See the packet dump from wireshark:
>> User Datagram Protocol, Src Port: na-localise (5062), Dst Port: sip (5060)
>>    Source port: na-localise (5062)
>>    Destination port: sip (5060)
>>    Length: 793
>>    Checksum: 0x34ff [validation disabled]
>> Session Initiation Protocol
>>    Request-Line: REGISTER sip:sip.test.com SIP/2.0
>>        Method: REGISTER
>>        Request-URI: sip:sip.test.com
>>        [Resent Packet: False]
>>    Message Header
>>        CSeq: 535 REGISTER
>>        Via: SIP/2.0/UDP 192.168.0.12:5062;branch=z9hG4bK2edb4d2b-d007-1910-9210-1cc1de60bc3c;rport
>>        User-Agent: TMSoftPhone
>>        [truncated] Authorization: Digest username="621", realm="sip.test.com", nonce="c4f876ae-eb58-11e1-8919-6df0d0a4f1d4", uri="sip:sip.test.com", algorithm=MD5, response="52625726279aaf004b61871164b61088", cnonce="0560cd01-d007-1910-91f
>>        From: <sip:621 at sip.test.com>;tag=25ffcc01-d007-1910-91fb-1cc1de60bc3c
>>        Call-ID: 25ffcc01-d007-1910-91fa-1cc1de60bc3c
>>        To: <sip:621 at sip.test.com>
>>        Contact: <sip:621 at 192.168.0.12:5062;transport=tcp>
>>        Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
>>        Expires: 120
>>        Content-Length: 0
>>        Max-Forwards: 70
>> 
>> ---
>> ??????? ???????? ??????????
>> ??????????? ????????
>> ??? "?????"
>> ???.   +7 495 7238000, ???. 101
>> ????.+7 495 7308111
>> E-mail: ananiev at svyaz.com
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lists at kavun.ch  Wed Aug 22 19:56:45 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 22 Aug 2012 11:56:45 -0400
Subject: [Freeswitch-users] Concatenated multiple playback types
In-Reply-To: <1FFF97C269757C458224B7C895F35F1514B3F0@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1514B3F0@cantor.std.visionutv.se>
Message-ID: <E114CDD1-849D-4621-8BCE-6DD9AC968EA3@kavun.ch>

Thanks for your response.
I don't think it would be something bad to implement. 

I also wanted to correct a little issueI experience both with FS and Asterisk.
If you generate a tone, even if it's played in a file, and you execute a record action right after it is played, the last 30 ms (or so) of the tone is distorded and kind of faded out.
In some sircompstances, you might even hear the hint of the tone when you execute the next playback.
I wanted to replace my record tones with the tone followed by 30 MS silence.

Best,
Emrah

On Aug 22, 2012, at 10:36 AM, Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> I don't believe this is possible. If you record 5 seconds of silence into a file you might be able to use file_string:// to play them both.
> 
> /Peter
> 
> 
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Emrah
> Skickat: den 22 augusti 2012 16:24
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] Concatenated multiple playback types
> 
> Hello there,
> 
> I am trying to add some silence before the alone sound file in my conference.cnf.xml and can't get it to work.
> 
> Here is what I tried:
> <param name="alone-sound" value="silence_stream://5000!file_string://conference/conf-alone.wav"/>
> <param name="alone-sound" value="silence_stream://5000&file_string://conference/conf-alone.wav"/>
> <param name="alone-sound" value="silence_stream://5000|file_string://conference/conf-alone.wav"/>
> <param name="alone-sound" value="silence_stream://5000;file_string://conference/conf-alone.wav"/>
> 
> Any idea on how to do this?
> 
> Thanks!
> Emrah
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
>  
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> !DSPAM:5034e9a632761303016055!
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 kris at kriskinc.com  Wed Aug 22 19:58:12 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 22 Aug 2012 11:58:12 -0400
Subject: [Freeswitch-users] Polycom firmware and TLS
Message-ID: <CAKDfjgeZe7opEJGwZTuJNhTt-uBHLaLfD-Ckx7HYRskk9chk4Q@mail.gmail.com>

All,

  Which versions of Polycom firmware have you had the most success
using TLS with?

Thanks!

-- 
Kristian Kielhofner


From kris at kriskinc.com  Wed Aug 22 20:00:05 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 22 Aug 2012 12:00:05 -0400
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <CD9593DE-1F02-4135-AEDD-7ED7C6A5FB22@visionutveckling.se>
References: <5034019b.c456b60a.3fd2.ffffdc5cSMTPIN_ADDED@mx.google.com>
	<CALNZuwmN+xfGkW0HqqDpXdJ4WaW5o59BcX-RiuKdNo18QH5D7g@mail.gmail.com>
	<CD9593DE-1F02-4135-AEDD-7ED7C6A5FB22@visionutveckling.se>
Message-ID: <CAKDfjgddj7S7D=0-MtAqqsocCxPMHB7rgDQTPrDBhJmKZuYdzw@mail.gmail.com>

+1

On Wed, Aug 22, 2012 at 2:12 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> If your current hardware doesn't support more RAM - buy a new server. FS scales well, but it will need the memory and CPU to perform.
>
> To redesign FS to not create a new thread per channel would be a major rewrite of the entire core.
>
> Considering your requirements it sonds like Kamailio or OpenSIPS is a better alternative for you, especially since you don't want to hande media anyway.
>
> /Peter
>
> 22 aug 2012 kl. 00:39 skrev "Jason Caulfield" <jason.caulfield at intermetro.net<mailto:jason.caulfield at intermetro.net>>:
>
> It is not so much the cps but the concurrent calls.
>
> I am looking at maintaining a 30 cps with a 5 min. avg. call length which results in a sustained 9000 concurrent calls.
>
> This results in 85% mem. usage on a 4 GB machine, However CPU is at 25% (total) on a dual core with hyper threading.
>
> Current hardware does not support increase memory.
>
> I am striving for about 75% cpu (appox. 90 cps).
>
> Jason
>
>
> From: 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 Avi Marcus
> Sent: Tuesday, August 21, 2012 2:58 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FreeSWITCH Question
>
> Do you have such a large cps/concurrent volume that you're actually seeing a performance hit?
>
> -Avi
>
> On Wed, Aug 22, 2012 at 12:35 AM, Jason Caulfield <jason.caulfield at intermetro.net<mailto:jason.caulfield at intermetro.net>> wrote:
> I am using FreeSWITCH in media bypass mode to decrease memory and cpu usage.
>
> I notice that mod_sofia creates threads for each call leg to maintain session context.
>
> Do you know of a way to configure FreeSWITCH to use a table to maintain the context when in media bypass mode to reduce the number of threads?
>
> I am hoping that this will speed things up by reducing thread context switching and reduce memory usage by decreasing memory allocation for each thread.
>
> Thanks for the help,
> Jason
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> 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:50340a7d32766392456629!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> 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:50340a7d32766392456629!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Kristian Kielhofner


From stargray at bigmir.net  Wed Aug 22 19:42:59 2012
From: stargray at bigmir.net (=?KOI8-R?B?4c7Uz84g98/KzMXOy88=?=)
Date: Wed, 22 Aug 2012 18:42:59 +0300
Subject: [Freeswitch-users] FreeSWITCH SIP trunk with CUCM 7.1
Message-ID: <CAHwAnSFyCQv_dELUKFe_jBmAYcvetw1uLOFbZ-cioZzCuZhi=w@mail.gmail.com>

Good day!

We have next task. We need config SIP trunk between FreeSWITCH and CUCM
7.1. FreeSWITCH have to register on CUCM.

We have made in CUCM SIP trunk with correct SIP Trunk Security Profile for
authorization and created Application User.

But FreeSWITCH can not registered. There is 405 error "SIP trunk disallow
REGISTER". See attachments.

Maybe someone faced with a similar problem?


Thanks,
Anton
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From msc at freeswitch.org  Wed Aug 22 20:04:32 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Aug 2012 09:04:32 -0700
Subject: [Freeswitch-users] Polycom firmware and TLS
In-Reply-To: <CAKDfjgeZe7opEJGwZTuJNhTt-uBHLaLfD-Ckx7HYRskk9chk4Q@mail.gmail.com>
References: <CAKDfjgeZe7opEJGwZTuJNhTt-uBHLaLfD-Ckx7HYRskk9chk4Q@mail.gmail.com>
Message-ID: <CAKzWOxU5c2YbhELsyKnUt=RFVod5bLqeRUUkVMdxoJEC_freHA@mail.gmail.com>

On Wed, Aug 22, 2012 at 8:58 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> All,
>
>   Which versions of Polycom firmware have you had the most success
> using TLS with?
>
> Thanks!
>

I'm interested in this as well. I believe our interop pages haven't had
much in the way of updates lately and all the major MFGs have released new
versions of their firmware. It would be nice to know what works and what
doesn't.
-MC
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From msc at freeswitch.org  Wed Aug 22 20:07:48 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Aug 2012 09:07:48 -0700
Subject: [Freeswitch-users] Concatenated multiple playback types
In-Reply-To: <E114CDD1-849D-4621-8BCE-6DD9AC968EA3@kavun.ch>
References: <1FFF97C269757C458224B7C895F35F1514B3F0@cantor.std.visionutv.se>
	<E114CDD1-849D-4621-8BCE-6DD9AC968EA3@kavun.ch>
Message-ID: <CAKzWOxWF-3MFy+09BjmJg4LiVnRdziZOidhWnYAUkLhGB+GkCg@mail.gmail.com>

You could always take the sneaky approach and use sox to add some silence
to the beginning of conf-alone.wav.
-MC

On Wed, Aug 22, 2012 at 8:56 AM, Emrah <lists at kavun.ch> wrote:

> Thanks for your response.
> I don't think it would be something bad to implement.
>
> I also wanted to correct a little issueI experience both with FS and
> Asterisk.
> If you generate a tone, even if it's played in a file, and you execute a
> record action right after it is played, the last 30 ms (or so) of the tone
> is distorded and kind of faded out.
> In some sircompstances, you might even hear the hint of the tone when you
> execute the next playback.
> I wanted to replace my record tones with the tone followed by 30 MS
> silence.
>
> Best,
> Emrah
>
> On Aug 22, 2012, at 10:36 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> > I don't believe this is possible. If you record 5 seconds of silence
> into a file you might be able to use file_string:// to play them both.
> >
> > /Peter
> >
> >
> > -----Ursprungligt meddelande-----
> > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] F?r Emrah
> > Skickat: den 22 augusti 2012 16:24
> > Till: FreeSWITCH Users Help
> > ?mne: [Freeswitch-users] Concatenated multiple playback types
> >
> > Hello there,
> >
> > I am trying to add some silence before the alone sound file in my
> conference.cnf.xml and can't get it to work.
> >
> > Here is what I tried:
> > <param name="alone-sound"
> value="silence_stream://5000!file_string://conference/conf-alone.wav"/>
> > <param name="alone-sound"
> value="silence_stream://5000&file_string://conference/conf-alone.wav"/>
> > <param name="alone-sound"
> value="silence_stream://5000|file_string://conference/conf-alone.wav"/>
> > <param name="alone-sound"
> value="silence_stream://5000;file_string://conference/conf-alone.wav"/>
> >
> > Any idea on how to do this?
> >
> > Thanks!
> > Emrah
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
> > !DSPAM:5034e9a632761303016055!
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From mike at jerris.com  Wed Aug 22 20:18:58 2012
From: mike at jerris.com (Michael Jerris)
Date: Wed, 22 Aug 2012 12:18:58 -0400
Subject: [Freeswitch-users] Build on Apple PPC?
In-Reply-To: <6B206226-751D-43F2-B214-614A0009BF38@insensate.co.uk>
References: <6B206226-751D-43F2-B214-614A0009BF38@insensate.co.uk>
Message-ID: <54B7795D-2F35-4D31-A364-E2663D68BA6E@jerris.com>

Please open a bug on this issue on jira.freeswitch.org in the "build system" component.  I was just looking at that section last night and it did look right so I will need to dig in to this further.  Please provide full log of the configure attached to the bug.

Thanks
Mike

On Aug 22, 2012, at 8:54 AM, Lawrence Conroy <lconroy at insensate.co.uk> wrote:

> Hi there,
> I'm trying to build 1.21 stable from git (replacing my current 1.06 tarball build) on a TiBook (PPC G4) running Leopard (10.5.8).
> 
> It's failing during configure -- "configure: error: no usable libodbc; please install unixodbc devel package or equivalent".
> I used the command line flag --disable-core-odbc-support, so don't understand ... why is configure looking for odbc at all?
> 
> anyone have an idea why this flag is ignored?



From mike at jerris.com  Wed Aug 22 20:22:40 2012
From: mike at jerris.com (Michael Jerris)
Date: Wed, 22 Aug 2012 12:22:40 -0400
Subject: [Freeswitch-users] Routing inbound calls to extentions with did
	intact
In-Reply-To: <6293150F220E44F5A0438104F2A46271@NDX0>
References: <6293150F220E44F5A0438104F2A46271@NDX0>
Message-ID: <7FFEE577-2E7A-46C5-BA15-35B0384C6283@jerris.com>

Check out http://wiki.freeswitch.org/wiki/Function_sofia_contact specifically the syntax using the ^.  Does this get you what you want, if not, how close is it?  We could add a similar syntax for other behavior.  Could you explain what exactly you want different in the sip packet?

On Aug 22, 2012, at 10:52 AM, Ian Halliday <ihalliday at ndevix.com> wrote:

> Hello All,
> 
> I use freeswitch to route inbound calls to some of the PBX's we host, and I
> use the following method (Below) to do so. It's been working great (20,000
> calls in the first month without a problem), but I wanted to get a second
> opinion to see if there was a better way to do such a thing (keeping the DID
> intact). Any help would be appreciated. (I used an example DID and extention)
> 
> <extension name="setdid">
> <condition field="destination_number" expression="^setdid$">
> <action application="set" data="dialed_ext=${did}"/>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set" data="disable_q850_reason=true"/>
> <action application="bridge"
> data="sofia/internal/${did}${regex(${sofia_contact(${user})}|^[^\@]+(.*)|%1)}"/>
> </condition>
> </extension>
> 
> <extension name="13125551234">
> <condition field="destination_number" expression="^13125551234$">
> <action application="set" data="user=user1002"/>
> <action application="set" data="did=${destination_number}"/>
> <action application="execute_extension" data="setdid"/>
> </condition>
> </extension>
> 

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From msc at freeswitch.org  Wed Aug 22 20:26:03 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Aug 2012 09:26:03 -0700
Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today
 (Important Information - Please Read)
Message-ID: <CAKzWOxX0_c41zoW7y3JHqP33tSV0xe4Pmcp_7ioddTdVMw_qvg@mail.gmail.com>

Hello all!

The conference will start in less than an hour. Here's our agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22

We've made a change to the system. Everyone will come in muted. When you
press 0 it will send a message into the #FreeSWITCH IRC channel - "user has
a question". One of the conference moderators will unmute you so that you
can ask your question. We are hoping this will make the conference more
enjoyable for everyone.

Talk to you soon!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From lists at kavun.ch  Wed Aug 22 20:27:07 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 22 Aug 2012 12:27:07 -0400
Subject: [Freeswitch-users] Concatenated multiple playback types
In-Reply-To: <CAKzWOxWF-3MFy+09BjmJg4LiVnRdziZOidhWnYAUkLhGB+GkCg@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F1514B3F0@cantor.std.visionutv.se>
	<E114CDD1-849D-4621-8BCE-6DD9AC968EA3@kavun.ch>
	<CAKzWOxWF-3MFy+09BjmJg4LiVnRdziZOidhWnYAUkLhGB+GkCg@mail.gmail.com>
Message-ID: <55F547E7-F986-4268-9BBD-C4210D997D12@kavun.ch>

Hey Michael,

That was my work around before I posted, but I thought since you guys do allow some flexibility with silence, tone generation and file playback all usable with the same app, I thought it would make more sense to take advantage of it and combine some things there.

All the best, thanks for your response.
Emrah
On Aug 22, 2012, at 12:07 PM, Michael Collins <msc at freeswitch.org> wrote:

> You could always take the sneaky approach and use sox to add some silence to the beginning of conf-alone.wav.
> -MC
> 
> On Wed, Aug 22, 2012 at 8:56 AM, Emrah <lists at kavun.ch> wrote:
> Thanks for your response.
> I don't think it would be something bad to implement.
> 
> I also wanted to correct a little issueI experience both with FS and Asterisk.
> If you generate a tone, even if it's played in a file, and you execute a record action right after it is played, the last 30 ms (or so) of the tone is distorded and kind of faded out.
> In some sircompstances, you might even hear the hint of the tone when you execute the next playback.
> I wanted to replace my record tones with the tone followed by 30 MS silence.
> 
> Best,
> Emrah
> 
> On Aug 22, 2012, at 10:36 AM, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> 
> > I don't believe this is possible. If you record 5 seconds of silence into a file you might be able to use file_string:// to play them both.
> >
> > /Peter
> >
> >
> > -----Ursprungligt meddelande-----
> > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Emrah
> > Skickat: den 22 augusti 2012 16:24
> > Till: FreeSWITCH Users Help
> > ?mne: [Freeswitch-users] Concatenated multiple playback types
> >
> > Hello there,
> >
> > I am trying to add some silence before the alone sound file in my conference.cnf.xml and can't get it to work.
> >
> > Here is what I tried:
> > <param name="alone-sound" value="silence_stream://5000!file_string://conference/conf-alone.wav"/>
> > <param name="alone-sound" value="silence_stream://5000&file_string://conference/conf-alone.wav"/>
> > <param name="alone-sound" value="silence_stream://5000|file_string://conference/conf-alone.wav"/>
> > <param name="alone-sound" value="silence_stream://5000;file_string://conference/conf-alone.wav"/>
> >
> > Any idea on how to do this?
> >
> > Thanks!
> > Emrah
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> >  
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
> > !DSPAM:5034e9a632761303016055!
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mike at jerris.com  Wed Aug 22 20:28:31 2012
From: mike at jerris.com (Michael Jerris)
Date: Wed, 22 Aug 2012 12:28:31 -0400
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <CAKDfjgddj7S7D=0-MtAqqsocCxPMHB7rgDQTPrDBhJmKZuYdzw@mail.gmail.com>
References: <5034019b.c456b60a.3fd2.ffffdc5cSMTPIN_ADDED@mx.google.com>
	<CALNZuwmN+xfGkW0HqqDpXdJ4WaW5o59BcX-RiuKdNo18QH5D7g@mail.gmail.com>
	<CD9593DE-1F02-4135-AEDD-7ED7C6A5FB22@visionutveckling.se>
	<CAKDfjgddj7S7D=0-MtAqqsocCxPMHB7rgDQTPrDBhJmKZuYdzw@mail.gmail.com>
Message-ID: <5FF9C3DB-85A4-4933-8528-0EA4C4585F4F@jerris.com>

if you paypal me $30 or so I can send you this thing that should roughly double your capacity:

http://www.memorydepot.com/list.asp?CAT=PC31333&capacity=4GB


On Aug 22, 2012, at 12:00 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:

> +1
> 
> On Wed, Aug 22, 2012 at 2:12 AM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
>> If your current hardware doesn't support more RAM - buy a new server. FS scales well, but it will need the memory and CPU to perform.
>> 
>> To redesign FS to not create a new thread per channel would be a major rewrite of the entire core.
>> 
>> Considering your requirements it sonds like Kamailio or OpenSIPS is a better alternative for you, especially since you don't want to hande media anyway.
>> 
>> /Peter
>> 
>> 22 aug 2012 kl. 00:39 skrev "Jason Caulfield" <jason.caulfield at intermetro.net<mailto:jason.caulfield at intermetro.net>>:
>> 
>> It is not so much the cps but the concurrent calls.
>> 
>> I am looking at maintaining a 30 cps with a 5 min. avg. call length which results in a sustained 9000 concurrent calls.
>> 
>> This results in 85% mem. usage on a 4 GB machine, However CPU is at 25% (total) on a dual core with hyper threading.
>> 
>> Current hardware does not support increase memory.
>> 
>> I am striving for about 75% cpu (appox. 90 cps).
>> 
>> Jason
>> 
>> 
>> From: 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 Avi Marcus
>> Sent: Tuesday, August 21, 2012 2:58 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] FreeSWITCH Question
>> 
>> Do you have such a large cps/concurrent volume that you're actually seeing a performance hit?
>> 
>> -Avi
>> 
>> On Wed, Aug 22, 2012 at 12:35 AM, Jason Caulfield <jason.caulfield at intermetro.net<mailto:jason.caulfield at intermetro.net>> wrote:
>> I am using FreeSWITCH in media bypass mode to decrease memory and cpu usage.
>> 
>> I notice that mod_sofia creates threads for each call leg to maintain session context.
>> 
>> Do you know of a way to configure FreeSWITCH to use a table to maintain the context when in media bypass mode to reduce the number of threads?
>> 
>> I am hoping that this will speed things up by reducing thread context switching and reduce memory usage by decreasing memory allocation for each thread.
>> 
>> Thanks for the help,
>> Jason
>> 

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From msc at freeswitch.org  Wed Aug 22 20:40:19 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Aug 2012 09:40:19 -0700
Subject: [Freeswitch-users] Concatenated multiple playback types
In-Reply-To: <55F547E7-F986-4268-9BBD-C4210D997D12@kavun.ch>
References: <1FFF97C269757C458224B7C895F35F1514B3F0@cantor.std.visionutv.se>
	<E114CDD1-849D-4621-8BCE-6DD9AC968EA3@kavun.ch>
	<CAKzWOxWF-3MFy+09BjmJg4LiVnRdziZOidhWnYAUkLhGB+GkCg@mail.gmail.com>
	<55F547E7-F986-4268-9BBD-C4210D997D12@kavun.ch>
Message-ID: <CAKzWOxVs9jfyn7Qjit-vJHnxKzLFbvrCkpzDNgHut0UrwRoj2A@mail.gmail.com>

One thing I like about the sox method is that it requires very little in
the way of processing/resources.
-MC

On Wed, Aug 22, 2012 at 9:27 AM, Emrah <lists at kavun.ch> wrote:

> Hey Michael,
>
> That was my work around before I posted, but I thought since you guys do
> allow some flexibility with silence, tone generation and file playback all
> usable with the same app, I thought it would make more sense to take
> advantage of it and combine some things there.
>
> All the best, thanks for your response.
> Emrah
> On Aug 22, 2012, at 12:07 PM, Michael Collins <msc at freeswitch.org> wrote:
>
> > You could always take the sneaky approach and use sox to add some
> silence to the beginning of conf-alone.wav.
> > -MC
> >
> > On Wed, Aug 22, 2012 at 8:56 AM, Emrah <lists at kavun.ch> wrote:
> > Thanks for your response.
> > I don't think it would be something bad to implement.
> >
> > I also wanted to correct a little issueI experience both with FS and
> Asterisk.
> > If you generate a tone, even if it's played in a file, and you execute a
> record action right after it is played, the last 30 ms (or so) of the tone
> is distorded and kind of faded out.
> > In some sircompstances, you might even hear the hint of the tone when
> you execute the next playback.
> > I wanted to replace my record tones with the tone followed by 30 MS
> silence.
> >
> > Best,
> > Emrah
> >
> > On Aug 22, 2012, at 10:36 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
> >
> > > I don't believe this is possible. If you record 5 seconds of silence
> into a file you might be able to use file_string:// to play them both.
> > >
> > > /Peter
> > >
> > >
> > > -----Ursprungligt meddelande-----
> > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] F?r Emrah
> > > Skickat: den 22 augusti 2012 16:24
> > > Till: FreeSWITCH Users Help
> > > ?mne: [Freeswitch-users] Concatenated multiple playback types
> > >
> > > Hello there,
> > >
> > > I am trying to add some silence before the alone sound file in my
> conference.cnf.xml and can't get it to work.
> > >
> > > Here is what I tried:
> > > <param name="alone-sound"
> value="silence_stream://5000!file_string://conference/conf-alone.wav"/>
> > > <param name="alone-sound"
> value="silence_stream://5000&file_string://conference/conf-alone.wav"/>
> > > <param name="alone-sound"
> value="silence_stream://5000|file_string://conference/conf-alone.wav"/>
> > > <param name="alone-sound"
> value="silence_stream://5000;file_string://conference/conf-alone.wav"/>
> > >
> > > Any idea on how to do this?
> > >
> > > Thanks!
> > > Emrah
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.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
> > >
> > > !DSPAM:5034e9a632761303016055!
> > >
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.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
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
> >
> >
> > --
> > Michael S Collins
> > Twitter: @mercutioviz
> > http://www.FreeSWITCH.org
> > http://www.ClueCon.com
> > http://www.OSTAG.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From alex at thewinelake.com  Wed Aug 22 22:11:39 2012
From: alex at thewinelake.com (Alex Lake)
Date: Wed, 22 Aug 2012 19:11:39 +0100
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <5FF9C3DB-85A4-4933-8528-0EA4C4585F4F@jerris.com>
References: <5034019b.c456b60a.3fd2.ffffdc5cSMTPIN_ADDED@mx.google.com>
	<CALNZuwmN+xfGkW0HqqDpXdJ4WaW5o59BcX-RiuKdNo18QH5D7g@mail.gmail.com>
	<CD9593DE-1F02-4135-AEDD-7ED7C6A5FB22@visionutveckling.se>
	<CAKDfjgddj7S7D=0-MtAqqsocCxPMHB7rgDQTPrDBhJmKZuYdzw@mail.gmail.com>
	<5FF9C3DB-85A4-4933-8528-0EA4C4585F4F@jerris.com>
Message-ID: <ABBB8AD8-B3CF-4A8C-8A5C-E85CE9168DE7@digitalmail.com>

Suspect that won't work. He's probably running it on. MacBook air or some such ;-)



On 22 Aug 2012, at 17:28, Michael Jerris <mike at jerris.com> wrote:

> if you paypal me $30 or so I can send you this thing that should roughly double your capacity:
> 
> http://www.memorydepot.com/list.asp?CAT=PC31333&capacity=4GB
> 
> 
> On Aug 22, 2012, at 12:00 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> 
>> +1
>> 
>> On Wed, Aug 22, 2012 at 2:12 AM, Peter Olsson
>> <peter.olsson at visionutveckling.se> wrote:
>>> If your current hardware doesn't support more RAM - buy a new server. FS scales well, but it will need the memory and CPU to perform.
>>> 
>>> To redesign FS to not create a new thread per channel would be a major rewrite of the entire core.
>>> 
>>> Considering your requirements it sonds like Kamailio or OpenSIPS is a better alternative for you, especially since you don't want to hande media anyway.
>>> 
>>> /Peter
>>> 
>>> 22 aug 2012 kl. 00:39 skrev "Jason Caulfield" <jason.caulfield at intermetro.net<mailto:jason.caulfield at intermetro.net>>:
>>> 
>>> It is not so much the cps but the concurrent calls.
>>> 
>>> I am looking at maintaining a 30 cps with a 5 min. avg. call length which results in a sustained 9000 concurrent calls.
>>> 
>>> This results in 85% mem. usage on a 4 GB machine, However CPU is at 25% (total) on a dual core with hyper threading.
>>> 
>>> Current hardware does not support increase memory.
>>> 
>>> I am striving for about 75% cpu (appox. 90 cps).
>>> 
>>> Jason
>>> 
>>> 
>>> From: 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 Avi Marcus
>>> Sent: Tuesday, August 21, 2012 2:58 PM
>>> To: FreeSWITCH Users Help
>>> Subject: Re: [Freeswitch-users] FreeSWITCH Question
>>> 
>>> Do you have such a large cps/concurrent volume that you're actually seeing a performance hit?
>>> 
>>> -Avi
>>> 
>>> On Wed, Aug 22, 2012 at 12:35 AM, Jason Caulfield <jason.caulfield at intermetro.net<mailto:jason.caulfield at intermetro.net>> wrote:
>>> I am using FreeSWITCH in media bypass mode to decrease memory and cpu usage.
>>> 
>>> I notice that mod_sofia creates threads for each call leg to maintain session context.
>>> 
>>> Do you know of a way to configure FreeSWITCH to use a table to maintain the context when in media bypass mode to reduce the number of threads?
>>> 
>>> I am hoping that this will speed things up by reducing thread context switching and reduce memory usage by decreasing memory allocation for each thread.
>>> 
>>> Thanks for the help,
>>> Jason
>>> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nickolayr at gmail.com  Wed Aug 22 22:56:32 2012
From: nickolayr at gmail.com (Nikolay Rogoshchenkov)
Date: Wed, 22 Aug 2012 14:56:32 -0400
Subject: [Freeswitch-users] FreeSWITCH SIP trunk with CUCM 7.1
In-Reply-To: <CAHwAnSFyCQv_dELUKFe_jBmAYcvetw1uLOFbZ-cioZzCuZhi=w@mail.gmail.com>
References: <CAHwAnSFyCQv_dELUKFe_jBmAYcvetw1uLOFbZ-cioZzCuZhi=w@mail.gmail.com>
Message-ID: <CAAQCs318Qwt+wY8b8GdgrhrH=syM6a9eONp+cuQNihkF-1xy+A@mail.gmail.com>

*"Warning: 399 ipcc-cucm1 "SIP trunk disallows REGISTER""*
- are you sure that you really need to register FS?

may be this link will be usefull for you:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/security/7_0_1/secugd/secrealm.html

--
Rogoshchenkov Nikolay


On Wed, Aug 22, 2012 at 11:42 AM, ????? ???????? <stargray at bigmir.net>wrote:

> Good day!
>
> We have next task. We need config SIP trunk between FreeSWITCH and CUCM
> 7.1. FreeSWITCH have to register on CUCM.
>
> We have made in CUCM SIP trunk with correct SIP Trunk Security Profilefor authorization and created Application User.
>
> But FreeSWITCH can not registered. There is 405 error "SIP trunk disallow
> REGISTER". See attachments.
>
> Maybe someone faced with a similar problem?
>
>
> Thanks,
> Anton
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lists at kavun.ch  Thu Aug 23 00:42:28 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 22 Aug 2012 16:42:28 -0400
Subject: [Freeswitch-users] Wrong energy level announcement in conference
Message-ID: <47B9F46D-89E7-40BF-86C6-ABCCC8692EA0@kavun.ch>

Hello,

My conference room is set with default energy-level = 300

The first time I use the 7 key, I hear "Zero", while in reality, I went from 300 to 100.

Where can we configure the adjustment levels and matching audio prompts? So to have zero for 0, one for 100, two for 200, and have each 7 or 9 key press change the value by 100?

Best to all,
Emrah

From lists at kavun.ch  Thu Aug 23 01:03:51 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 22 Aug 2012 17:03:51 -0400
Subject: [Freeswitch-users] Wideband Skype to SIP?
Message-ID: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>

Hi guys,

I am looking for a SIP based solution to interconnect with Skype, with wideband codec support if possible.

Does that exist? 
Otherwise, does Skypopen run on Debian Squeeze?

Best,
Emrah

From bdfoster at endigotech.com  Thu Aug 23 03:07:37 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 22 Aug 2012 19:07:37 -0400
Subject: [Freeswitch-users] Wideband Skype to SIP?
In-Reply-To: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>
References: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>
Message-ID: <CAKV-jFwM7TKt6HKaKfD4HEG+9KAvDh1J=ZLP+W9dwnVQfb6_Yg@mail.gmail.com>

Mod_skypopen works well on Debian Squeeze.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 22, 2012 5:05 PM, "Emrah" <lists at kavun.ch> wrote:

> Hi guys,
>
> I am looking for a SIP based solution to interconnect with Skype, with
> wideband codec support if possible.
>
> Does that exist?
> Otherwise, does Skypopen run on Debian Squeeze?
>
> Best,
> Emrah
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at endigotech.com  Thu Aug 23 03:38:16 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 22 Aug 2012 19:38:16 -0400
Subject: [Freeswitch-users] Polycom firmware and TLS
In-Reply-To: <CAKzWOxU5c2YbhELsyKnUt=RFVod5bLqeRUUkVMdxoJEC_freHA@mail.gmail.com>
References: <CAKDfjgeZe7opEJGwZTuJNhTt-uBHLaLfD-Ckx7HYRskk9chk4Q@mail.gmail.com>
	<CAKzWOxU5c2YbhELsyKnUt=RFVod5bLqeRUUkVMdxoJEC_freHA@mail.gmail.com>
Message-ID: <CAKV-jFwYbr9O=EH=_kQsUOqOnY0xHRqyBn_SmHTcQkC-pkyxZw@mail.gmail.com>

I have a phone running 4.0.2 ish Polycom UC firmware. I might have to go
ahead and bite the bullet and test. I don't have much experience in the way
of TLS, so it's time to pull my finger out and start learning.

- BDF

On Wed, Aug 22, 2012 at 12:04 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
>  On Wed, Aug 22, 2012 at 8:58 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:
>
>> All,
>>
>>   Which versions of Polycom firmware have you had the most success
>> using TLS with?
>>
>> Thanks!
>>
>
> I'm interested in this as well. I believe our interop pages haven't had
> much in the way of updates lately and all the major MFGs have released new
> versions of their firmware. It would be nice to know what works and what
> doesn't.
> -MC
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From bdfoster at endigotech.com  Thu Aug 23 03:43:27 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 22 Aug 2012 19:43:27 -0400
Subject: [Freeswitch-users] Wideband Skype to SIP?
In-Reply-To: <CAKV-jFwM7TKt6HKaKfD4HEG+9KAvDh1J=ZLP+W9dwnVQfb6_Yg@mail.gmail.com>
References: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>
	<CAKV-jFwM7TKt6HKaKfD4HEG+9KAvDh1J=ZLP+W9dwnVQfb6_Yg@mail.gmail.com>
Message-ID: <CAKV-jFy2jzH99S4u9G+KL1WkWUQPC1rUEMV7pRLcL7sL-yLUyA@mail.gmail.com>

P.S. you might want to check this out:

http://www.skype.com/intl/en-us/business/skype-connect#t_pricing

It doesn't mention if you can do wideband with skype users, but you can't
do wideband when you connect to the PSTN anyways. Sounds like you might be
able to get away with this free if all you want to do is to connect with
your skype users. Like I said before, mod_skypopen works well with Debian
Squeeze, and we do use it here.

Take it easy,

Brian Foster
Endigo Computer LLC

On Wed, Aug 22, 2012 at 7:07 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> Mod_skypopen works well on Debian Squeeze.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
>  On Aug 22, 2012 5:05 PM, "Emrah" <lists at kavun.ch> wrote:
>
>> Hi guys,
>>
>> I am looking for a SIP based solution to interconnect with Skype, with
>> wideband codec support if possible.
>>
>> Does that exist?
>> Otherwise, does Skypopen run on Debian Squeeze?
>>
>> Best,
>> Emrah
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From lists at kavun.ch  Thu Aug 23 04:07:35 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 22 Aug 2012 20:07:35 -0400
Subject: [Freeswitch-users] Wideband Skype to SIP?
In-Reply-To: <CAKV-jFy2jzH99S4u9G+KL1WkWUQPC1rUEMV7pRLcL7sL-yLUyA@mail.gmail.com>
References: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>
	<CAKV-jFwM7TKt6HKaKfD4HEG+9KAvDh1J=ZLP+W9dwnVQfb6_Yg@mail.gmail.com>
	<CAKV-jFy2jzH99S4u9G+KL1WkWUQPC1rUEMV7pRLcL7sL-yLUyA@mail.gmail.com>
Message-ID: <93FD4ECA-6F9A-4C74-8ACE-1FBC8BF405CF@kavun.ch>

I think I'll try mod_skypopen. Do you have any how to that lists the appropriate dependencies for Debian?

Thanks and regards,
Emrah
--
Emrah

?In theory, theory and practice are the same. In practice, they are not.? Albert Einstein

On Aug 22, 2012, at 7:43 PM, Brian Foster <bdfoster at endigotech.com> wrote:

> P.S. you might want to check this out:
>  
> http://www.skype.com/intl/en-us/business/skype-connect#t_pricing
>  
> It doesn't mention if you can do wideband with skype users, but you can't do wideband when you connect to the PSTN anyways. Sounds like you might be able to get away with this free if all you want to do is to connect with your skype users. Like I said before, mod_skypopen works well with Debian Squeeze, and we do use it here.
>  
> Take it easy,
>  
> Brian Foster
> Endigo Computer LLC
> 
> On Wed, Aug 22, 2012 at 7:07 PM, Brian Foster <bdfoster at endigotech.com> wrote:
> Mod_skypopen works well on Debian Squeeze.
> 
> Brian Foster
> Endigo Computer LLC
> 
> Sent from a mobile device.
> 
> On Aug 22, 2012 5:05 PM, "Emrah" <lists at kavun.ch> wrote:
> Hi guys,
> 
> I am looking for a SIP based solution to interconnect with Skype, with wideband codec support if possible.
> 
> Does that exist?
> Otherwise, does Skypopen run on Debian Squeeze?
> 
> Best,
> Emrah
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> 
> -- 
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
> 
> This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lists at kavun.ch  Thu Aug 23 04:20:24 2012
From: lists at kavun.ch (Emrah)
Date: Wed, 22 Aug 2012 20:20:24 -0400
Subject: [Freeswitch-users] Dial out within a conference with early media
Message-ID: <EC5B3D9E-68E3-4974-AA17-EFE65D96CC52@kavun.ch>

Hi guys,

Is there an option to dial with early media support from within a conference?

This is what happens now:

1. I have an active conf. 
2. I decide to pull in an additional user by calling her mobile.
3. From fs_cli, I execute conference <conf-id> dial Sofia/gateway/xyz/1234567890. Then
4. Either she picks up, we her the enter sound and we know she's joined in.
5. Or her mobile is switched off, with no voicemail and just an INFO message... And we will never know what happened.

Is there a way to accomplish the following?

Have the conference broadcast the early media portion of the call, and still play the enter sound when the called party picks up?

If this is not possible, I think it would be a great addition to the conferencing module of FS to have a dedicated dial command with support for Early Media.

Best and thanks,
Emrah
--
Emrah

?In theory, theory and practice are the same. In practice, they are not.? Albert Einstein

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From asaad2 at gmail.com  Thu Aug 23 05:44:13 2012
From: asaad2 at gmail.com (BookBag)
Date: Wed, 22 Aug 2012 21:44:13 -0400
Subject: [Freeswitch-users] FreeSWITCH SIP trunk with CUCM 7.1
In-Reply-To: <CAHwAnSFyCQv_dELUKFe_jBmAYcvetw1uLOFbZ-cioZzCuZhi=w@mail.gmail.com>
References: <CAHwAnSFyCQv_dELUKFe_jBmAYcvetw1uLOFbZ-cioZzCuZhi=w@mail.gmail.com>
Message-ID: <CAAdTMa-Tjrxq6UUcosFN=f7pNPXqTZXnu3tL_uf6ChAwQ5ezRQ@mail.gmail.com>

I actually have that running. I configured a sip trunk between fs and cucm
7.1.3.  The trick is avoid sip authentication at all costs. Cucm does not
do sip authentication. It can neither accept or register to another sip
trunk.

Two things need to be done. First on cucm create a sip trunk to fs using ip
address and in your dialplan establish a sip dial route with a prefix like
for example 81. So all cisco users who dial 81xxxx will be routed through
that trunk to your fs ipaddress.

The other thing that you do is edit your acl to accept everything from your
cucm ip address. Then in your dial plan establish another prefix for
example for example 71. So all fs users will dial 71xxxx to get to a cucm
extension will go directly to xxxx@<cucm ip>.  Let me know if you need
additional help.
On Aug 22, 2012 12:03 PM, "????? ????????" <stargray at bigmir.net> wrote:

> Good day!
>
> We have next task. We need config SIP trunk between FreeSWITCH and CUCM
> 7.1. FreeSWITCH have to register on CUCM.
>
> We have made in CUCM SIP trunk with correct SIP Trunk Security Profilefor authorization and created Application User.
>
> But FreeSWITCH can not registered. There is 405 error "SIP trunk disallow
> REGISTER". See attachments.
>
> Maybe someone faced with a similar problem?
>
>
> Thanks,
> Anton
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 dvl36.ripe.nick at gmail.com  Thu Aug 23 08:13:02 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Thu, 23 Aug 2012 07:13:02 +0300
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAA2gtHAfQBGPeUxt+Ckko84+NCwOyD=wocFvOUgA+Op7hUeu=w@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
	<CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
	<CAA2gtHBXf-RfZ4eFB1k7NfCaYccqU6uK78NnxxBL0C=dO2PPBA@mail.gmail.com>
	<CAMjaK3DCzKEi7Gep9eZLhGs4FGix3edCDCxtymJXCwWHP9FSjA@mail.gmail.com>
	<CAA2gtHAcp_2+ch0-TOK-6XibCdmdFzfWOvJ9s2_7JKLjnf7MTA@mail.gmail.com>
	<CAMjaK3D+3XQDwU3s8m-FUj0wK_vPjOir3n3Lrvfwq9dS3Je2WQ@mail.gmail.com>
	<CAA2gtHBHkmUWM6EyqP14RtoH-jLUWitUdHXkZYKAVp3e54eBmw@mail.gmail.com>
	<CAA2gtHCpqRFB+d8xE5DWYbuC2U8uaWV_56VUzpSh7t2zzUvtHQ@mail.gmail.com>
	<CAMjaK3A0aKF80nFCm-roRB1MC65RA3Rt-0QErx=6py8L4K7r_w@mail.gmail.com>
	<CAA2gtHA5M1AqdsxSgwWQfXASymCS0qO6QEXXH6+aWddP2UgTOA@mail.gmail.com>
	<CAMjaK3BFRXcKrBXHz2Z7zxOTm-AW2XfkN2BBXDNLFhQgQ1Csvw@mail.gmail.com>
	<CAA2gtHAfQBGPeUxt+Ckko84+NCwOyD=wocFvOUgA+Op7hUeu=w@mail.gmail.com>
Message-ID: <CAMjaK3AhbVuZq=8aCrMtPbb31zPKKwfQheS0zgOybfsh2a8_gQ@mail.gmail.com>

OK.

May be it depends of modem firmware, never seen before E1550 with 4 ttyUSB.
wvdial method is not precise for searching right modem control device port.
I suggest to use picocom/minicom.
Make sure that not gsmopen, nor any other software that may use serial port
is running.
Run:
# ls /dev/ttyUSB*
/dev/ttyUSB0 /dev/ttyUSB1 /dev/ttyUSB2

Connect to serial port (recommend  revers order)

#picocom /dev/ttyUSBx
then call to modem sim-card number.
If you will see:

RING

That port will be controldevice_name.
For controldevice_audio_name use x-1, usually it work.
Last of /dev/ttyUSB usually is controldevice_name.

Best Wishes,
Dmitry.

2012/8/22 Sameer Khan <sameer2k3t at gmail.com>

> It worked with
>
>     <param name="controldevice_name" value="/dev/ttyUSB3"/>
>       <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>
>
> I tried usermod -a -G dialout root
>
> before typo as freeswitch was running with root, so may it fixed the
> issue.
>
> Thanks alot for your help my friend
>
>
>
> On Wed, Aug 22, 2012 at 6:49 PM, Dmitry Lysenko <dvl36.ripe.nick at gmail.com
> > wrote:
>
>> So, if it doesn't work with audio device at /dev/ttyUSB2, try /dev/ttyUSB1
>> Typo was at ttyUSB1 too.
>>
>> 2012/8/22 Sameer Khan <sameer2k3t at gmail.com>
>>
>>> Could not be
>>> because freeswitch is being run as root
>>>
>>> And it can load /dev/ttyUSB0
>>>
>>> sorry I had typo in audio line, It can load ttyUSB2 now
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Aug 22, 2012 at 6:03 PM, Dmitry Lysenko <
>>> dvl36.ripe.nick at gmail.com> wrote:
>>>
>>>> Maybe:
>>>>
>>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/087009.html
>>>> ?
>>>>
>>>> 2012/8/22 Sameer Khan <sameer2k3t at gmail.com>
>>>>
>>>>> this is wvdial output
>>>>>
>>>>>
>>>>> Found a modem on /dev/ttyUSB0.
>>>>> Modem configuration written to /etc/wvdial.conf.
>>>>> ttyUSB0<Info>: Speed 9600; init "ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0"
>>>>> ttyUSB3<Info>: Speed 9600; init "ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0"
>>>>>
>>>>>
>>>>> On Wed, Aug 22, 2012 at 5:39 PM, Sameer Khan <sameer2k3t at gmail.com>wrote:
>>>>>
>>>>>> No luck
>>>>>>
>>>>>> 2012-08-22 17:35:38.337967 [ERR] gsmopen_protocol.cpp:3160 rev
>>>>>> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
>>>>>> ttyUSB2, NOT open
>>>>>>
>>>>>> 2012-08-22 17:37:27.057967 [ERR] gsmopen_protocol.cpp:3160 rev
>>>>>> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  3160 ][gsm01     ][-1, 0, 0] port
>>>>>> ttyUSB1, NOT open
>>>>>>
>>>>>> Yes It is enabled, I tested before on other box
>>>>>>
>>>>>>
>>>>>> On Wed, Aug 22, 2012 at 5:01 PM, Dmitry Lysenko <
>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> I have reproduced  such situation. It seems that
>>>>>>> controldevice_audio_name is pointing to wrong /dev/ttyUSBx.
>>>>>>> I suggest try:
>>>>>>>
>>>>>>>  <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>>>>
>>>>>>> if doesn't help try:
>>>>>>>  <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>>>>>>>
>>>>>>> P.S. Are you sure modem voice feature enabled?
>>>>>>>
>>>>>>> 2012/8/22 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>
>>>>>>>> Hi,
>>>>>>>>
>>>>>>>> It says
>>>>>>>>
>>>>>>>> option                  8690  0
>>>>>>>> usb_wwan                9960  1 option
>>>>>>>> usbserial              31372  2 option,usb_wwan
>>>>>>>>
>>>>>>>> What should I do now to get it working ?
>>>>>>>>
>>>>>>>> Thanks for replying
>>>>>>>>
>>>>>>>> On Tue, Aug 21, 2012 at 2:55 PM, Dmitry Lysenko <
>>>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> As root: lsmod | grep option
>>>>>>>>> Should be something like that:
>>>>>>>>> ---
>>>>>>>>> root at debian:~# lsmod | grep option
>>>>>>>>> option                 15351  6
>>>>>>>>> usb_wwan                8362  1 option
>>>>>>>>> usbserial              26535  14 usb_wwan,option
>>>>>>>>> usbcore               122921  10
>>>>>>>>> ehci_hcd,usb_storage,usbserial,gspca_main,usb_wwan,option,gspca_zc3xx,snd_usbmidi_lib,snd_usb_audio
>>>>>>>>> ---
>>>>>>>>>  2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>>>
>>>>>>>>>> I guess so, I have CentOS 6 minimal installed
>>>>>>>>>>
>>>>>>>>>> how can I identify if this is option driver ?
>>>>>>>>>>
>>>>>>>>>> Yes I saw this message on ttyUSB0
>>>>>>>>>>
>>>>>>>>>> I can not load gsmopen with ttyUSB2 or ttyUSB1
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Tue, Aug 21, 2012 at 2:21 PM, Dmitry Lysenko <
>>>>>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Are you using "option" linux driver?
>>>>>>>>>>> I checked Huawei E1550, E153 and E173 modems on some systems and
>>>>>>>>>>> they have only 3 virtual serial ports.(using "option" driver)
>>>>>>>>>>> Also you can check manually with minicom.
>>>>>>>>>>> Usually ttyUSB0 is accepting  AT-commands, but will not reply
>>>>>>>>>>> with OK or ERROR, ttyUSB1 is used for audio, and ttyUSB2 is accepting
>>>>>>>>>>> AT-commands and replying  as normal serial modem.
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> 2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>>>>>
>>>>>>>>>>>> It didn't for me, according to wvdial my modem is ttyUSB0 and
>>>>>>>>>>>> ttyUSB3
>>>>>>>>>>>>
>>>>>>>>>>>> On Mon, Aug 20, 2012 at 1:43 PM, Dmitry Lysenko <
>>>>>>>>>>>> dvl36.ripe.nick at gmail.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> Hello.
>>>>>>>>>>>>> Try:
>>>>>>>>>>>>>  <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>>>>>>>>>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>>>>>>>>>> It's worked for me.
>>>>>>>>>>>>>
>>>>>>>>>>>>> 2012/8/20 Sameer Khan <sameer2k3t at gmail.com>
>>>>>>>>>>>>>
>>>>>>>>>>>>>>  Hi,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I can't get audio working with e1550
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I have this config :
>>>>>>>>>>>>>>     <interface id="1" name="gsm01">
>>>>>>>>>>>>>>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>>>>>>>>>>>       <param name="controldevice_audio_name"
>>>>>>>>>>>>>> value="/dev/ttyUSB3"/>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I tried swaping 0 and 3 but having same results
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I am originating a call from softphone and answering it on
>>>>>>>>>>>>>> cell phone but softphone doesn't get notified of being answered and keeps
>>>>>>>>>>>>>> ringing
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Please help
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Thanks
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> 
>>>>>>>>>>>>>> 
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>>>
>>>>>>>>>>>>> 
>>>>>>>>>>>>> 
>>>>>>>>>>>>>
>>>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>>
>>>>>>>>>>>> 
>>>>>>>>>>>> 
>>>>>>>>>>>>
>>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>>
>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 dvl36.ripe.nick at gmail.com  Thu Aug 23 08:51:53 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Thu, 23 Aug 2012 07:51:53 +0300
Subject: [Freeswitch-users] mod_gsmopen: ROAMING Network ERROR
In-Reply-To: <CALXCt0q+CZsOBL6Uv0_W9TMje+qZGb+rguej-cgMqRNQA9mo6w@mail.gmail.com>
References: <CAD=Wdqi_rbz3Z0XrYcS_CP29wSV94G8R7cJpHKRqZoeNorhmxw@mail.gmail.com>
	<CAMjaK3B-Y+nswvVnj49Jh_+sZCJKvcXiSmhJ7ksMWo6UXbGt9Q@mail.gmail.com>
	<CALXCt0q+CZsOBL6Uv0_W9TMje+qZGb+rguej-cgMqRNQA9mo6w@mail.gmail.com>
Message-ID: <CAMjaK3AEPEc8BbTanipJ0s4dOO2qgg2xAkK6pYkaU8-bLSG8mg@mail.gmail.com>

Hi Giovanni

Thank you for good software. I used asterisk and chan_dongle  before, but
freeswitch and gsmopen is leaving them behind.
As for wiki, sorry but my english is not good enough.
I have some questions about gsmopen functionality. But after your holidays.

Thank you.

2012/8/21 Giovanni Maruzzelli <gmaruzz at gmail.com>

> hi Sameer,
> I'm in holidays.
>
> If problems persists, please contact me after Sept 10th.
>
> Dimitri, you're right. Btw, you seems very knowledgeable about gsmopen
> and related stuff. Please feel free to modify expand mod-gsmopen wiki
> page and/or send patches to jira. Thanks in advance Dimitri!
>
> -giovanni
>
> On 8/21/12, Dmitry Lysenko <dvl36.ripe.nick at gmail.com> wrote:
> > Hello.
> > I have the same "ROAMING Network" ALERTs, it's annoying, but everything
> > working OK.
> > Checked with  Huawei E1550.
> >
> > P.S. Are you sure that your Huawei E169  is voice enabled?
> >
> > 2012/8/21 Josep Maria Vi?olas <vinolasbcn at gmail.com>
> >
> >> I have a Huawei E169 with a SIM from a "virtual provider" that uses
> other
> >> provider's networks, so it is allways technically 'roaming'.
> >>
> >> With mod_gsmopen the problem is that throws an error about "ROAMING
> >> Network" and also an ALERT. I don't know if that is why it is not able
> to
> >> make calls. In that case, how can I tell the module that roaming is not
> a
> >> bad thing??
> >>
> >> This is freeswitch debug output when I load mod_gsmmobile:
> >> ------------------------------------------
> >>
> > ...
> >
> >> ------------------------------------------
> >> ** From extension 1000 I hear a calling tone even the Freeswitch log
> says
> >> that ATD+N? command failed.
> >>
> >> Anyone knows if the problem is the ROAMING Network error??
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 dvl36.ripe.nick at gmail.com  Thu Aug 23 08:59:39 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Thu, 23 Aug 2012 07:59:39 +0300
Subject: [Freeswitch-users] mod_gsmopen serial init failed
In-Reply-To: <20120821111808.210950@gmx.com>
References: <20120821111808.210950@gmx.com>
Message-ID: <CAMjaK3DNsO+X81kNS4yb3tKerEmAnuC1UwxS=xmPyNmXEAsUcQ@mail.gmail.com>

If gsmopen doesn't work well with /dev/ttyUSB0 as control device try use
/dev/ttyUSB2.

2012/8/21 Mi Ke <mi.ke at null.net>

> It appears that selinux policies prevented freeswitch user from accessing
> /dev/ttyUSB*
>
> [root at freeswitch ~]# ls -l /dev/ttyUSB*
> crw-rw----. 1 root dialout 188, 0 Aug 21 10:57 /dev/ttyUSB0
> crw-rw----. 1 root dialout 188, 1 Aug 21 10:48 /dev/ttyUSB1
> crw-rw----. 1 root dialout 188, 2 Aug 21 10:48 /dev/ttyUSB2
>
> so adding freeswitch user to dialout group by running "usermod -a -G
> dialout freeswitch" worked for me.
>
> Have a nice day
> Mike
>
>
>
>  ----- Original Message -----
>
> From: Mi Ke
>
> Sent: 08/21/12 12:58 PM
>
> To: FreeSWITCH Users Help
>
> Subject: [Freeswitch-users] mod_gsmopen serial init failed
>
>  Hi All,
>
> I have problems getting mod_gsmopen to see my E1550. Both minicom and
> wvdial see the dongle on /dev/ttyUSB0:
>
> ttyUSB0<*1>: ATQ0 V1 E1 -- OK
> ttyUSB0<*1>: ATQ0 V1 E1 Z -- OK
> ttyUSB0<*1>: ATQ0 V1 E1 S0=0 -- OK
> ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 -- OK
> ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 -- OK
> ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0 -- OK
> ttyUSB0<*1>: Modem Identifier: ATI -- Manufacturer: huawei
> ttyUSB0<*1>: Speed 9600: AT -- OK
> ttyUSB0<*1>: Max speed is 9600; that should be safe.
> ttyUSB0<*1>: ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0 -- OK
>
> Welcome to minicom 2.3
>
> OPTIONS: I18n
> Compiled on Aug 19 2010, 05:50:19.
> Port /dev/ttyUSB0
>
> Press CTRL-A Z for help on special keys
> AT S7=45 S0=0 L1 V1 X4 &c1 E1 Q0
> OK
>
> at+cpin?
> +CPIN: READY
> OK
>
> but mod_gsmopen does not want to see it:
>
> 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1203 rev
> ccae5cd|07bc7ba[(nil)|37     ][NOTICA  1203 ][none      ][-1,-1,-1] GSMOPEN
> Charset Output Test 0 ?????
> 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1204 rev
> ccae5cd|07bc7ba[(nil)|37     ][NOTICA  1204 ][none      ][-1,-1,-1] GSMOPEN
> Charset Output Test 1 ???^?
> 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1205 rev
> ccae5cd|07bc7ba[(nil)|37     ][NOTICA  1205 ][none      ][-1,-1,-1] GSMOPEN
> Charset Output Test 2 ???????
> 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1206 rev
> ccae5cd|07bc7ba[(nil)|37     ][NOTICA  1206 ][none      ][-1,-1,-1] GSMOPEN
> Charset Output Test 3 ??????
> 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1207 rev
> ccae5cd|07bc7ba[(nil)|37     ][NOTICA  1207 ][none      ][-1,-1,-1] GSMOPEN
> Charset Output Test 4 ???
> 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1208 rev
> ccae5cd|07bc7ba[(nil)|37     ][NOTICA  1208 ][none      ][-1,-1,-1] GSMOPEN
> Charset Output Test 5 ??
> 2012-08-21 09:18:18.973501 [NOTICE] mod_gsmopen.cpp:1209 rev
> ccae5cd|07bc7ba[(nil)|37     ][NOTICA  1209 ][none      ][-1,-1,-1] GSMOPEN
> Charset Output Test 6 ??
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1226 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1226 ][none      ][-1,-1,-1]
> globals.debug=0
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1228 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1228 ][none      ][-1,-1,-1]
> globals.debug=8
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1234 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1234 ][none      ][-1,-1,-1]
> globals.dialplan=XML
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1240 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1240 ][none      ][-1,-1,-1]
> globals.context=external
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1231 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1231 ][none      ][-1,-1,-1]
> globals.hold_music=
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1237 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1237 ][none      ][-1,-1,-1]
> globals.destination=5000
> 2012-08-21 09:18:18.973501 [WARNING] mod_gsmopen.cpp:1653 rev
> ccae5cd|07bc7ba[(nil)|37     ][WARNINGA  1653 ][gsm00     ][-1, 0, 0]
> STARTING interface_id=1
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1654 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1654 ][gsm00     ][-1, 0, 0]
> id=1
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1655 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1655 ][gsm00     ][-1, 0, 0]
> name=gsm00
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1656 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1656 ][gsm00     ][-1, 0, 0]
> hold-music=
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1657 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1657 ][gsm00     ][-1, 0, 0]
> context=default
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1658 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1658 ][gsm00     ][-1, 0, 0]
> dialplan=XML
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1659 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1659 ][gsm00     ][-1, 0, 0]
> destination=5000
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1660 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1660 ][gsm00     ][-1, 0, 0]
> controldevice_name=/dev/ttyUSB0
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1661 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1661 ][gsm00     ][-1, 0, 0]
> controldevice_audio_name=/dev/ttyUSB2
> 2012-08-21 09:18:18.973501 [DEBUG] mod_gsmopen.cpp:1663 rev
> ccae5cd|07bc7ba[(nil)|37     ][DEBUG_GSMOPEN  1663 ][gsm00     ][-1, 0, 0]
> gsmopen_serial_sync_period=300
> 2012-08-21 09:18:18.973501 [ERR] gsmopen_protocol.cpp:122 rev
> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  122  ][gsm00     ][-1, 0, 0] port
> /dev/ttyUSB0, NOT open
> 2012-08-21 09:18:18.973501 [ERR] mod_gsmopen.cpp:1670 rev
> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  1670 ][gsm00     ][-1, 0, 0]
> gsmopen_serial_init failed
> 2012-08-21 09:18:18.973501 [ERR] mod_gsmopen.cpp:1671 rev
> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  1671 ][gsm00     ][-1, 0, 0]
> STARTING interface_id=1 FAILED
> 2012-08-21 09:18:18.973501 [ERR] mod_gsmopen.cpp:2928 rev
> ccae5cd|07bc7ba[(nil)|37     ][ERRORA  2928 ][gsm00     ][-1, 0, 0] ALARM
> on interface gsm00:
> 2012-08-21 09:18:18.973501 [CONSOLE] switch_loadable_module.c:1328
> Successfully Loaded [mod_gsmopen]
> 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:146 Adding
> Endpoint 'gsmopen'
> 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding
> API Function 'gsm'
> 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding
> API Function 'gsmopen'
> 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding
> API Function 'gsmopen_boost_audio'
> 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding
> API Function 'gsmopen_dump'
> 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:298 Adding
> API Function 'gsmopen_sendsms'
> 2012-08-21 09:18:18.973501 [NOTICE] switch_loadable_module.c:403 Adding
> Chat interface 'sms'
>
> Dongle is voice enabled, PIN request is disabled...What do I do wrong ?
>
> Cheers / Mike
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 dvl36.ripe.nick at gmail.com  Thu Aug 23 10:11:18 2012
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Thu, 23 Aug 2012 09:11:18 +0300
Subject: [Freeswitch-users] Gsmopen : no audio in huawei e1550
In-Reply-To: <CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
References: <CAA2gtHAiqKKsAFBU5j8bvoT_M9vxbmnsPX2MAEVAWacAU7_yAw@mail.gmail.com>
	<CAMjaK3CO9qz401MpBCHWSvdcPpzzBiWPDM6xPD2nAMor1XCnKA@mail.gmail.com>
	<CAA2gtHBVLmAobtq56mc5xEHgdakAiyA++rJJUAh8dKbrChop1Q@mail.gmail.com>
	<CAMjaK3C3HGXb7G=Ay0gy6t9mB0ixPqgvge76FqNvvzN9Tw+e9w@mail.gmail.com>
Message-ID: <CAMjaK3ANw=e+ni53evnBV07XG710brcHZWYrY-csPQcnKpxrbA@mail.gmail.com>

I should correct this:
----
>Usually ttyUSB0 is accepting  AT-commands, but will not reply with OK or
ERROR, ttyUSB1 is used for >audio, and ttyUSB2 is accepting AT-commands and
replying  as normal serial modem.
----
I checked other E1550 modem and it responds to AT-commands with OK/ERROR on
both serial ports (first and last), but messages indicating external
events,such as "RING", will appear only on one of this ports. This port
should be used for control device, usually it has bigger number. It seems
that it depend of modem firmware version.

Best regards,
 Dmitry.

2012/8/21 Dmitry Lysenko <dvl36.ripe.nick at gmail.com>

> Are you using "option" linux driver?
> I checked Huawei E1550, E153 and E173 modems on some systems and they have
> only 3 virtual serial ports.(using "option" driver)
> Also you can check manually with minicom.
> Usually ttyUSB0 is accepting  AT-commands, but will not reply with OK or
> ERROR, ttyUSB1 is used for audio, and ttyUSB2 is accepting AT-commands and
> replying  as normal serial modem.
>
>
> 2012/8/21 Sameer Khan <sameer2k3t at gmail.com>
>
>> It didn't for me, according to wvdial my modem is ttyUSB0 and ttyUSB3
>>
>> On Mon, Aug 20, 2012 at 1:43 PM, Dmitry Lysenko <
>> dvl36.ripe.nick at gmail.com> wrote:
>>
>>>
>>> Hello.
>>> Try:
>>>  <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>  <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>> It's worked for me.
>>>
>>> 2012/8/20 Sameer Khan <sameer2k3t at gmail.com>
>>>
>>>>  Hi,
>>>>
>>>> I can't get audio working with e1550
>>>>
>>>> I have this config :
>>>>     <interface id="1" name="gsm01">
>>>>       <param name="controldevice_name" value="/dev/ttyUSB0"/>
>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB3"/>
>>>>
>>>> I tried swaping 0 and 3 but having same results
>>>>
>>>> I am originating a call from softphone and answering it on cell phone
>>>> but softphone doesn't get notified of being answered and keeps ringing
>>>>
>>>> Please help
>>>>
>>>> Thanks
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 tahir at ictinnovations.com  Thu Aug 23 11:14:03 2012
From: tahir at ictinnovations.com (tahir almas)
Date: Thu, 23 Aug 2012 12:14:03 +0500
Subject: [Freeswitch-users] FreeSWITCH Auto Dialer - Open source
In-Reply-To: <CAJUJwtg7vDr+MDbTvXJJk4HRAr38CTOYY3mRpWwBui0erBXLeg@mail.gmail.com>
References: <CA+aeUL_D-1sgqxvxshU9oqELh66Ck-_kTxesMCCjnv3DgDu-yg@mail.gmail.com>
	<502D0CAC.8030905@gmail.com>
	<CAJUJwtg7vDr+MDbTvXJJk4HRAr38CTOYY3mRpWwBui0erBXLeg@mail.gmail.com>
Message-ID: <CAOL0t+LwiTzxXY_VtA6FfN34NzQZYNcBHTHiFfrCK3ZyJMsvYw@mail.gmail.com>

I will recommend to check ICTDialer http://www.ictdialer.org that is based
on Plivo Communication framework / Freeswitch and support for standard
voice broadcasting as well as Interactive Voice broadcasting also you have
support IVR Designer to design your IVR with drag and drop feature

Regards

*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT




On Thu, Aug 16, 2012 at 11:41 PM, SamyGo <govoiper at gmail.com> wrote:

>
> Hi,
>
> Well I'd say to figure out your requirements first. I once got into same
> situation as your's and believe me reinventing the wheel (tailor made
> solution) for exactly the requirement was the best thing. Atleast I know
> I've created something which is doing all that I needed and the time spent
> on this was less than a week !!
>
> Its upto you and your requirements to choose the solution.
>
> BR
> Sammy Go.
> On Aug 16, 2012 8:11 PM, "Abaci" <abaci64 at gmail.com> wrote:
>
>>  http://www.ictdialer.org/
>>
>> On 8/16/2012 9:53 AM, Obie Brown wrote:
>>
>> Hi,
>>
>>  I really want to work with an Open Source dialer and FreeSWITCH but the
>> only thing I can find for this is Newfies Dialer which does not have
>> features of a Auto Dialer as I need. Also its use for Voice Broadcasting is
>> limited.
>>
>>  My question is. What other Auto Dialers that run on FreeSWITCH are
>> there. If any.
>>
>>
>>  Thanks,
>> Obie Brown
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> 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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 tahir at ictinnovations.com  Thu Aug 23 11:19:44 2012
From: tahir at ictinnovations.com (tahir almas)
Date: Thu, 23 Aug 2012 12:19:44 +0500
Subject: [Freeswitch-users] Voice broadcasting
In-Reply-To: <CAAdTMa-wwTiyZ+=caecU8mC5K8Z2A_2Atnq4yonDRz-172BU5g@mail.gmail.com>
References: <CAAdTMa-Qz1UyH+wXKEGXRBHF=Hjh7S_jpxe_BSphXgsrYpo_aw@mail.gmail.com>
	<CC359325.3A8AE%krice@freeswitch.org>
	<CAKV-jFzQizHQAbUEQxCPdi_Wc9odhcL0j7oc9grHX=kceZ3-aQ@mail.gmail.com>
	<CAAdTMa-wwTiyZ+=caecU8mC5K8Z2A_2Atnq4yonDRz-172BU5g@mail.gmail.com>
Message-ID: <CAOL0t+LYhOBid83cWPwvUEPqyE309Vm6DC7ZkoQWO42CVUp4Og@mail.gmail.com>

I recommend to check ICTDialer that has simpler design and based on Plivo /
Freeswitch http://www.ictdialer.org also supporting IVR Designer to draw
your custom outbound IVR with drag and drop feature  for Interactive Voice
Broadcasting , ICTDialer also support SMS Messaging and Fax broadcasting /
blasting

Regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT

****************************************************************************************************************
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On Wed, Jul 25, 2012 at 11:50 PM, BookBag <asaad2 at gmail.com> wrote:

> thanks everyone, I'll give it a try
>
>
> On Wed, Jul 25, 2012 at 1:11 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> Is newfies dialer a good fit for this case?
>>
>> Brian Foster
>> Endigo Computer LLC
>>
>> Sent from a mobile device.
>> On Jul 25, 2012 1:14 PM, "Ken Rice" <krice at freeswitch.org> wrote:
>>
>>>  You need a queue manager and all sorts of other instrumentation that
>>> is outside the scope of FreeSWITCH itself.
>>>
>>> The short answer, yes you can do it, the not so short answer, theres a
>>> lot of work to do to turn FreeSWITCH into a dialer.
>>>
>>> You might dig around in google to find some things that do robocalls on
>>> freeswitch already...
>>>
>>>
>>> On 7/25/12 11:56 AM, "BookBag" <asaad2 at gmail.com> wrote:
>>>
>>> thanx lol :-)
>>>
>>> But is there a way to have it call from a per-determined list
>>>
>>> On Wed, Jul 25, 2012 at 11:48 AM, Michael Collins <msc at freeswitch.org>
>>> wrote:
>>>
>>> >From the fs_cli:
>>> originate {ignore_early_media=true}sofia/gateway/your_gw_name/
>>> 18005551212 <tel:18005551212>  &playback("/path/to/file.wav")
>>>
>>> Please don't abuse your new found power! :)
>>> -MC
>>>
>>>
>>> On Wed, Jul 25, 2012 at 9:39 AM, BookBag <asaad2 at gmail.com> wrote:
>>>
>>> Hello all, can anyone direct to some information on how I can setup a
>>> voice broadcast on FS. I tried looking in the wiki but it justs redirects
>>> you to heading "Robocalls", which in turn redirect you "voice broadcast"
>>> but there is no information under it.
>>>
>>> Basically I just want to call about 100 people or so and play a
>>> recording and then hangup.
>>>
>>>
>>> If anyone could help me , i'd really appreciate it.
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> ------------------------------
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From stargray at bigmir.net  Thu Aug 23 11:38:08 2012
From: stargray at bigmir.net (Anton Vojlenko)
Date: Thu, 23 Aug 2012 10:38:08 +0300
Subject: [Freeswitch-users] FreeSWITCH SIP trunk with CUCM 7.1
In-Reply-To: <CAAdTMa-Tjrxq6UUcosFN=f7pNPXqTZXnu3tL_uf6ChAwQ5ezRQ@mail.gmail.com>
References: <CAHwAnSFyCQv_dELUKFe_jBmAYcvetw1uLOFbZ-cioZzCuZhi=w@mail.gmail.com>
	<CAAdTMa-Tjrxq6UUcosFN=f7pNPXqTZXnu3tL_uf6ChAwQ5ezRQ@mail.gmail.com>
Message-ID: <CAHwAnSEWoZyxBosP=MVjVctzi3_wAioUsv3_9kZEhD_yoWomJA@mail.gmail.com>

We have two CUCMs and I need create failover with them. How can I do this?

2012/8/23 BookBag <asaad2 at gmail.com>

> I actually have that running. I configured a sip trunk between fs and cucm
> 7.1.3.  The trick is avoid sip authentication at all costs. Cucm does not
> do sip authentication. It can neither accept or register to another sip
> trunk.
>
> Two things need to be done. First on cucm create a sip trunk to fs using
> ip address and in your dialplan establish a sip dial route with a prefix
> like for example 81. So all cisco users who dial 81xxxx will be routed
> through that trunk to your fs ipaddress.
>
> The other thing that you do is edit your acl to accept everything from
> your cucm ip address. Then in your dial plan establish another prefix for
> example for example 71. So all fs users will dial 71xxxx to get to a cucm
> extension will go directly to xxxx@<cucm ip>.  Let me know if you need
> additional help.
> On Aug 22, 2012 12:03 PM, "????? ????????" <stargray at bigmir.net> wrote:
>
>> Good day!
>>
>> We have next task. We need config SIP trunk between FreeSWITCH and CUCM
>> 7.1. FreeSWITCH have to register on CUCM.
>>
>> We have made in CUCM SIP trunk with correct SIP Trunk Security Profilefor authorization and created Application User.
>>
>> But FreeSWITCH can not registered. There is 405 error "SIP trunk disallow
>> REGISTER". See attachments.
>>
>> Maybe someone faced with a similar problem?
>>
>>
>> Thanks,
>> Anton
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 avi at avimarcus.net  Thu Aug 23 12:15:02 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 23 Aug 2012 11:15:02 +0300
Subject: [Freeswitch-users] FreeSWITCH Auto Dialer - Open source
In-Reply-To: <CAOL0t+LwiTzxXY_VtA6FfN34NzQZYNcBHTHiFfrCK3ZyJMsvYw@mail.gmail.com>
References: <CA+aeUL_D-1sgqxvxshU9oqELh66Ck-_kTxesMCCjnv3DgDu-yg@mail.gmail.com>
	<502D0CAC.8030905@gmail.com>
	<CAJUJwtg7vDr+MDbTvXJJk4HRAr38CTOYY3mRpWwBui0erBXLeg@mail.gmail.com>
	<CAOL0t+LwiTzxXY_VtA6FfN34NzQZYNcBHTHiFfrCK3ZyJMsvYw@mail.gmail.com>
Message-ID: <CALNZuwnkQ1+nNHECNOM6hs8xCTV1wx3-ucevp01ur8bW5Km1tQ@mail.gmail.com>

The demo seems broken: http://demo.ictdialer.org/

-Avi


On Thu, Aug 23, 2012 at 10:14 AM, tahir almas <tahir at ictinnovations.com>wrote:

> I will recommend to check ICTDialer http://www.ictdialer.org that is
> based on Plivo Communication framework / Freeswitch and support for
> standard voice broadcasting as well as Interactive Voice broadcasting also
> you have support IVR Designer to design your IVR with drag and drop feature
>
> Regards
>
> *Tahir Almas*
>
> Managing Partner
> ICT Innovations
> http://www.ictinnovations.com
> Leveraging open source in ICT
>
>
>
>
>
> On Thu, Aug 16, 2012 at 11:41 PM, SamyGo <govoiper at gmail.com> wrote:
>
>>
>> Hi,
>>
>> Well I'd say to figure out your requirements first. I once got into same
>> situation as your's and believe me reinventing the wheel (tailor made
>> solution) for exactly the requirement was the best thing. Atleast I know
>> I've created something which is doing all that I needed and the time spent
>> on this was less than a week !!
>>
>> Its upto you and your requirements to choose the solution.
>>
>> BR
>> Sammy Go.
>> On Aug 16, 2012 8:11 PM, "Abaci" <abaci64 at gmail.com> wrote:
>>
>>>  http://www.ictdialer.org/
>>>
>>> On 8/16/2012 9:53 AM, Obie Brown wrote:
>>>
>>> Hi,
>>>
>>>  I really want to work with an Open Source dialer and FreeSWITCH but
>>> the only thing I can find for this is Newfies Dialer which does not have
>>> features of a Auto Dialer as I need. Also its use for Voice Broadcasting is
>>> limited.
>>>
>>>  My question is. What other Auto Dialers that run on FreeSWITCH are
>>> there. If any.
>>>
>>>
>>>  Thanks,
>>> Obie Brown
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> 
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>
>>> 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
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 obiebrown at gmail.com  Thu Aug 23 12:20:25 2012
From: obiebrown at gmail.com (Obie Brown)
Date: Thu, 23 Aug 2012 17:50:25 +0930
Subject: [Freeswitch-users] FreeSWITCH Auto Dialer - Open source
In-Reply-To: <CALNZuwnkQ1+nNHECNOM6hs8xCTV1wx3-ucevp01ur8bW5Km1tQ@mail.gmail.com>
References: <CA+aeUL_D-1sgqxvxshU9oqELh66Ck-_kTxesMCCjnv3DgDu-yg@mail.gmail.com>
	<502D0CAC.8030905@gmail.com>
	<CAJUJwtg7vDr+MDbTvXJJk4HRAr38CTOYY3mRpWwBui0erBXLeg@mail.gmail.com>
	<CAOL0t+LwiTzxXY_VtA6FfN34NzQZYNcBHTHiFfrCK3ZyJMsvYw@mail.gmail.com>
	<CALNZuwnkQ1+nNHECNOM6hs8xCTV1wx3-ucevp01ur8bW5Km1tQ@mail.gmail.com>
Message-ID: <CA+aeUL9LUZjmJp=LwG2NijHXcZvU3ib+1DTVh1MkPjQapt2Ksw@mail.gmail.com>

Yes I was trying to look at demo the there day (few days ago) and there was
error. Still seems to be error.

*PDOException*: SQLSTATE[42S02]: Base table or view not found: 1146 Table
'call4smile.semaphore' doesn't exist: SELECT expire, value FROM {semaphore}
WHERE name = :name; Array ( [:name] => variable_init ) in *
lock_may_be_available()* (line *167* of*/usr/ictbilling/includes/lock.inc*).


If you can fix tahir is this something you can fix?


Thanks,
Obie Brown


On Thu, Aug 23, 2012 at 5:45 PM, Avi Marcus <avi at avimarcus.net> wrote:

> The demo seems broken: http://demo.ictdialer.org/
>
> -Avi
>
>
> On Thu, Aug 23, 2012 at 10:14 AM, tahir almas <tahir at ictinnovations.com>wrote:
>
>> I will recommend to check ICTDialer http://www.ictdialer.org that is
>> based on Plivo Communication framework / Freeswitch and support for
>> standard voice broadcasting as well as Interactive Voice broadcasting also
>> you have support IVR Designer to design your IVR with drag and drop feature
>>
>> Regards
>>
>> *Tahir Almas*
>>
>> Managing Partner
>> ICT Innovations
>> http://www.ictinnovations.com
>> Leveraging open source in ICT
>>
>>
>>
>>
>>
>> On Thu, Aug 16, 2012 at 11:41 PM, SamyGo <govoiper at gmail.com> wrote:
>>
>>>
>>> Hi,
>>>
>>> Well I'd say to figure out your requirements first. I once got into same
>>> situation as your's and believe me reinventing the wheel (tailor made
>>> solution) for exactly the requirement was the best thing. Atleast I know
>>> I've created something which is doing all that I needed and the time spent
>>> on this was less than a week !!
>>>
>>> Its upto you and your requirements to choose the solution.
>>>
>>> BR
>>> Sammy Go.
>>> On Aug 16, 2012 8:11 PM, "Abaci" <abaci64 at gmail.com> wrote:
>>>
>>>>  http://www.ictdialer.org/
>>>>
>>>> On 8/16/2012 9:53 AM, Obie Brown wrote:
>>>>
>>>> Hi,
>>>>
>>>>  I really want to work with an Open Source dialer and FreeSWITCH but
>>>> the only thing I can find for this is Newfies Dialer which does not have
>>>> features of a Auto Dialer as I need. Also its use for Voice Broadcasting is
>>>> limited.
>>>>
>>>>  My question is. What other Auto Dialers that run on FreeSWITCH are
>>>> there. If any.
>>>>
>>>>
>>>>  Thanks,
>>>> Obie Brown
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>>
>>>> 
>>>>
>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>>
>>>> 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
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 tpe at actimizer.com  Thu Aug 23 12:42:59 2012
From: tpe at actimizer.com (Tor Petterson)
Date: Thu, 23 Aug 2012 10:42:59 +0200
Subject: [Freeswitch-users] FreeSWITCH and ISDN
Message-ID: <CADnUQS_eejGVz3a47EX2jO9d5C7zf-o03HTuimi3Sbm=kPJrSw@mail.gmail.com>

Hi

I have a FreeSWITCH installation, doing outbound telephony, with a
peak load of about 500 to 600 concurrent channels.
Now I have to do a similar setup in a market where SIP telephony is
not available, so it has to be ISDN.
I am looking at buying either Digium or Sangoma cards. Does anyone
have any experiences about reliability of these cards?
How does the cpu load per channel compare to SIP telephony?

-- 
Tor Petterson

tpe at actimizer.com

Tobaksvejen 25, 2. tv. - 2860 S?borg
Telephone: +45 39 55 05 32
www.actimizer.com


From petedao at gmail.com  Thu Aug 23 14:14:36 2012
From: petedao at gmail.com (Pete Kay)
Date: Thu, 23 Aug 2012 18:14:36 +0800
Subject: [Freeswitch-users] No voice heard
Message-ID: <CAPxXe8XFonrLKFkHQZ4kx0r4Z-8PHX2rqyq978CRDa+Ghf_PdA@mail.gmail.com>

Hi,
I tried the following command.
<action application="playback"
data="${http_get(http://myserver.yo/media/hello_world.wav)}"/>

After I replaced the url to a correct URL, file can be downloaded but
no voice is heard.

Any idea how to troubleshoot this?

p


From bdfoster at endigotech.com  Thu Aug 23 15:07:36 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 23 Aug 2012 07:07:36 -0400
Subject: [Freeswitch-users] FreeSWITCH Auto Dialer - Open source
In-Reply-To: <CA+aeUL9LUZjmJp=LwG2NijHXcZvU3ib+1DTVh1MkPjQapt2Ksw@mail.gmail.com>
References: <CA+aeUL_D-1sgqxvxshU9oqELh66Ck-_kTxesMCCjnv3DgDu-yg@mail.gmail.com>
	<502D0CAC.8030905@gmail.com>
	<CAJUJwtg7vDr+MDbTvXJJk4HRAr38CTOYY3mRpWwBui0erBXLeg@mail.gmail.com>
	<CAOL0t+LwiTzxXY_VtA6FfN34NzQZYNcBHTHiFfrCK3ZyJMsvYw@mail.gmail.com>
	<CALNZuwnkQ1+nNHECNOM6hs8xCTV1wx3-ucevp01ur8bW5Km1tQ@mail.gmail.com>
	<CA+aeUL9LUZjmJp=LwG2NijHXcZvU3ib+1DTVh1MkPjQapt2Ksw@mail.gmail.com>
Message-ID: <CAKV-jFxr0b8sJoEto1PBonLGuqQh5GM5GWZkv+5EuO=VRz0AMQ@mail.gmail.com>

They might be having issues with their database. Simple error nothing to
worry about.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 23, 2012 4:25 AM, "Obie Brown" <obiebrown at gmail.com> wrote:

> Yes I was trying to look at demo the there day (few days ago) and there
> was error. Still seems to be error.
>
> *PDOException*: SQLSTATE[42S02]: Base table or view not found: 1146 Table
> 'call4smile.semaphore' doesn't exist: SELECT expire, value FROM {semaphore}
> WHERE name = :name; Array ( [:name] => variable_init ) in *
> lock_may_be_available()* (line *167* of*/usr/ictbilling/includes/lock.inc*
> ).
>
> If you can fix tahir is this something you can fix?
>
>
> Thanks,
> Obie Brown
>
>
> On Thu, Aug 23, 2012 at 5:45 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> The demo seems broken: http://demo.ictdialer.org/
>>
>> -Avi
>>
>>
>> On Thu, Aug 23, 2012 at 10:14 AM, tahir almas <tahir at ictinnovations.com>wrote:
>>
>>> I will recommend to check ICTDialer http://www.ictdialer.org that is
>>> based on Plivo Communication framework / Freeswitch and support for
>>> standard voice broadcasting as well as Interactive Voice broadcasting also
>>> you have support IVR Designer to design your IVR with drag and drop feature
>>>
>>> Regards
>>>
>>> *Tahir Almas*
>>>
>>> Managing Partner
>>> ICT Innovations
>>> http://www.ictinnovations.com
>>> Leveraging open source in ICT
>>>
>>>
>>>
>>>
>>>
>>> On Thu, Aug 16, 2012 at 11:41 PM, SamyGo <govoiper at gmail.com> wrote:
>>>
>>>>
>>>> Hi,
>>>>
>>>> Well I'd say to figure out your requirements first. I once got into
>>>> same situation as your's and believe me reinventing the wheel (tailor made
>>>> solution) for exactly the requirement was the best thing. Atleast I know
>>>> I've created something which is doing all that I needed and the time spent
>>>> on this was less than a week !!
>>>>
>>>> Its upto you and your requirements to choose the solution.
>>>>
>>>> BR
>>>> Sammy Go.
>>>> On Aug 16, 2012 8:11 PM, "Abaci" <abaci64 at gmail.com> wrote:
>>>>
>>>>>  http://www.ictdialer.org/
>>>>>
>>>>> On 8/16/2012 9:53 AM, Obie Brown wrote:
>>>>>
>>>>> Hi,
>>>>>
>>>>>  I really want to work with an Open Source dialer and FreeSWITCH but
>>>>> the only thing I can find for this is Newfies Dialer which does not have
>>>>> features of a Auto Dialer as I need. Also its use for Voice Broadcasting is
>>>>> limited.
>>>>>
>>>>>  My question is. What other Auto Dialers that run on FreeSWITCH are
>>>>> there. If any.
>>>>>
>>>>>
>>>>>  Thanks,
>>>>> Obie Brown
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>>>
>>>>> 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
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at endigotech.com  Thu Aug 23 15:09:08 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 23 Aug 2012 07:09:08 -0400
Subject: [Freeswitch-users] FreeSWITCH SIP trunk with CUCM 7.1
In-Reply-To: <CAHwAnSEWoZyxBosP=MVjVctzi3_wAioUsv3_9kZEhD_yoWomJA@mail.gmail.com>
References: <CAHwAnSFyCQv_dELUKFe_jBmAYcvetw1uLOFbZ-cioZzCuZhi=w@mail.gmail.com>
	<CAAdTMa-Tjrxq6UUcosFN=f7pNPXqTZXnu3tL_uf6ChAwQ5ezRQ@mail.gmail.com>
	<CAHwAnSEWoZyxBosP=MVjVctzi3_wAioUsv3_9kZEhD_yoWomJA@mail.gmail.com>
Message-ID: <CAKV-jFzU+908ZpxC0ap29UJ=-rfrQ93fUB_aTMfYHNYL8LGO6Q@mail.gmail.com>

Use OpenSIPS.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 23, 2012 3:39 AM, "Anton Vojlenko" <stargray at bigmir.net> wrote:

> We have two CUCMs and I need create failover with them. How can I do this?
>
> 2012/8/23 BookBag <asaad2 at gmail.com>
>
>> I actually have that running. I configured a sip trunk between fs and
>> cucm 7.1.3.  The trick is avoid sip authentication at all costs. Cucm does
>> not do sip authentication. It can neither accept or register to another sip
>> trunk.
>>
>> Two things need to be done. First on cucm create a sip trunk to fs using
>> ip address and in your dialplan establish a sip dial route with a prefix
>> like for example 81. So all cisco users who dial 81xxxx will be routed
>> through that trunk to your fs ipaddress.
>>
>> The other thing that you do is edit your acl to accept everything from
>> your cucm ip address. Then in your dial plan establish another prefix for
>> example for example 71. So all fs users will dial 71xxxx to get to a cucm
>> extension will go directly to xxxx@<cucm ip>.  Let me know if you need
>> additional help.
>> On Aug 22, 2012 12:03 PM, "????? ????????" <stargray at bigmir.net> wrote:
>>
>>> Good day!
>>>
>>> We have next task. We need config SIP trunk between FreeSWITCH and CUCM
>>> 7.1. FreeSWITCH have to register on CUCM.
>>>
>>> We have made in CUCM SIP trunk with correct SIP Trunk Security Profilefor authorization and created Application User.
>>>
>>> But FreeSWITCH can not registered. There is 405 error "SIP trunk
>>> disallow REGISTER". See attachments.
>>>
>>> Maybe someone faced with a similar problem?
>>>
>>>
>>> Thanks,
>>> Anton
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 avi at avimarcus.net  Thu Aug 23 15:27:26 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 23 Aug 2012 14:27:26 +0300
Subject: [Freeswitch-users] FreeSWITCH Auto Dialer - Open source
In-Reply-To: <CAKV-jFxr0b8sJoEto1PBonLGuqQh5GM5GWZkv+5EuO=VRz0AMQ@mail.gmail.com>
References: <CA+aeUL_D-1sgqxvxshU9oqELh66Ck-_kTxesMCCjnv3DgDu-yg@mail.gmail.com>
	<502D0CAC.8030905@gmail.com>
	<CAJUJwtg7vDr+MDbTvXJJk4HRAr38CTOYY3mRpWwBui0erBXLeg@mail.gmail.com>
	<CAOL0t+LwiTzxXY_VtA6FfN34NzQZYNcBHTHiFfrCK3ZyJMsvYw@mail.gmail.com>
	<CALNZuwnkQ1+nNHECNOM6hs8xCTV1wx3-ucevp01ur8bW5Km1tQ@mail.gmail.com>
	<CA+aeUL9LUZjmJp=LwG2NijHXcZvU3ib+1DTVh1MkPjQapt2Ksw@mail.gmail.com>
	<CAKV-jFxr0b8sJoEto1PBonLGuqQh5GM5GWZkv+5EuO=VRz0AMQ@mail.gmail.com>
Message-ID: <CALNZuwnv9RwLS2kBwc9x5NMg9TWzvo496FrpXFnPk0nNAxn7Zg@mail.gmail.com>

Brian Foster I'm not worried about it, I'd like to see the demo...
-Avi


On Thu, Aug 23, 2012 at 2:07 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> They might be having issues with their database. Simple error nothing to
> worry about.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Aug 23, 2012 4:25 AM, "Obie Brown" <obiebrown at gmail.com> wrote:
>
>> Yes I was trying to look at demo the there day (few days ago) and there
>> was error. Still seems to be error.
>>
>> *PDOException*: SQLSTATE[42S02]: Base table or view not found: 1146
>> Table 'call4smile.semaphore' doesn't exist: SELECT expire, value FROM
>> {semaphore} WHERE name = :name; Array ( [:name] => variable_init ) in *
>> lock_may_be_available()* (line *167* of*/usr/ictbilling/includes/lock.inc
>> *).
>>
>> If you can fix tahir is this something you can fix?
>>
>>
>> Thanks,
>> Obie Brown
>>
>>
>> On Thu, Aug 23, 2012 at 5:45 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> The demo seems broken: http://demo.ictdialer.org/
>>>
>>> -Avi
>>>
>>>
>>> On Thu, Aug 23, 2012 at 10:14 AM, tahir almas <tahir at ictinnovations.com>wrote:
>>>
>>>> I will recommend to check ICTDialer http://www.ictdialer.org that is
>>>> based on Plivo Communication framework / Freeswitch and support for
>>>> standard voice broadcasting as well as Interactive Voice broadcasting also
>>>> you have support IVR Designer to design your IVR with drag and drop feature
>>>>
>>>> Regards
>>>>
>>>> *Tahir Almas*
>>>>
>>>> Managing Partner
>>>> ICT Innovations
>>>> http://www.ictinnovations.com
>>>> Leveraging open source in ICT
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, Aug 16, 2012 at 11:41 PM, SamyGo <govoiper at gmail.com> wrote:
>>>>
>>>>>
>>>>> Hi,
>>>>>
>>>>> Well I'd say to figure out your requirements first. I once got into
>>>>> same situation as your's and believe me reinventing the wheel (tailor made
>>>>> solution) for exactly the requirement was the best thing. Atleast I know
>>>>> I've created something which is doing all that I needed and the time spent
>>>>> on this was less than a week !!
>>>>>
>>>>> Its upto you and your requirements to choose the solution.
>>>>>
>>>>> BR
>>>>> Sammy Go.
>>>>> On Aug 16, 2012 8:11 PM, "Abaci" <abaci64 at gmail.com> wrote:
>>>>>
>>>>>>  http://www.ictdialer.org/
>>>>>>
>>>>>> On 8/16/2012 9:53 AM, Obie Brown wrote:
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>>  I really want to work with an Open Source dialer and FreeSWITCH but
>>>>>> the only thing I can find for this is Newfies Dialer which does not have
>>>>>> features of a Auto Dialer as I need. Also its use for Voice Broadcasting is
>>>>>> limited.
>>>>>>
>>>>>>  My question is. What other Auto Dialers that run on FreeSWITCH are
>>>>>> there. If any.
>>>>>>
>>>>>>
>>>>>>  Thanks,
>>>>>> Obie Brown
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>>>>
>>>>>> 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
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 obiebrown at gmail.com  Thu Aug 23 15:39:12 2012
From: obiebrown at gmail.com (Obie Brown)
Date: Thu, 23 Aug 2012 21:09:12 +0930
Subject: [Freeswitch-users] FreeSWITCH Auto Dialer - Open source
In-Reply-To: <CALNZuwnv9RwLS2kBwc9x5NMg9TWzvo496FrpXFnPk0nNAxn7Zg@mail.gmail.com>
References: <CA+aeUL_D-1sgqxvxshU9oqELh66Ck-_kTxesMCCjnv3DgDu-yg@mail.gmail.com>
	<502D0CAC.8030905@gmail.com>
	<CAJUJwtg7vDr+MDbTvXJJk4HRAr38CTOYY3mRpWwBui0erBXLeg@mail.gmail.com>
	<CAOL0t+LwiTzxXY_VtA6FfN34NzQZYNcBHTHiFfrCK3ZyJMsvYw@mail.gmail.com>
	<CALNZuwnkQ1+nNHECNOM6hs8xCTV1wx3-ucevp01ur8bW5Km1tQ@mail.gmail.com>
	<CA+aeUL9LUZjmJp=LwG2NijHXcZvU3ib+1DTVh1MkPjQapt2Ksw@mail.gmail.com>
	<CAKV-jFxr0b8sJoEto1PBonLGuqQh5GM5GWZkv+5EuO=VRz0AMQ@mail.gmail.com>
	<CALNZuwnv9RwLS2kBwc9x5NMg9TWzvo496FrpXFnPk0nNAxn7Zg@mail.gmail.com>
Message-ID: <CA+aeUL8kFrwZR=worjfsozAv0Vs+jYLSDGFezvBVQhK3C8Gxcw@mail.gmail.com>

Yes looks very simple error. When its back I will check demo.

Thanks everyone for there suggestions.


Thanks,
Obie Brown


On Thu, Aug 23, 2012 at 8:57 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Brian Foster I'm not worried about it, I'd like to see the demo...
> -Avi
>
>
> On Thu, Aug 23, 2012 at 2:07 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> They might be having issues with their database. Simple error nothing to
>> worry about.
>>
>> Brian Foster
>> Endigo Computer LLC
>>
>> Sent from a mobile device.
>> On Aug 23, 2012 4:25 AM, "Obie Brown" <obiebrown at gmail.com> wrote:
>>
>>> Yes I was trying to look at demo the there day (few days ago) and there
>>> was error. Still seems to be error.
>>>
>>> *PDOException*: SQLSTATE[42S02]: Base table or view not found: 1146
>>> Table 'call4smile.semaphore' doesn't exist: SELECT expire, value FROM
>>> {semaphore} WHERE name = :name; Array ( [:name] => variable_init ) in *
>>> lock_may_be_available()* (line *167* of*
>>> /usr/ictbilling/includes/lock.inc*).
>>>
>>> If you can fix tahir is this something you can fix?
>>>
>>>
>>> Thanks,
>>> Obie Brown
>>>
>>>
>>> On Thu, Aug 23, 2012 at 5:45 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> The demo seems broken: http://demo.ictdialer.org/
>>>>
>>>> -Avi
>>>>
>>>>
>>>> On Thu, Aug 23, 2012 at 10:14 AM, tahir almas <tahir at ictinnovations.com
>>>> > wrote:
>>>>
>>>>> I will recommend to check ICTDialer http://www.ictdialer.org that is
>>>>> based on Plivo Communication framework / Freeswitch and support for
>>>>> standard voice broadcasting as well as Interactive Voice broadcasting also
>>>>> you have support IVR Designer to design your IVR with drag and drop feature
>>>>>
>>>>> Regards
>>>>>
>>>>> *Tahir Almas*
>>>>>
>>>>> Managing Partner
>>>>> ICT Innovations
>>>>> http://www.ictinnovations.com
>>>>> Leveraging open source in ICT
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Aug 16, 2012 at 11:41 PM, SamyGo <govoiper at gmail.com> wrote:
>>>>>
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> Well I'd say to figure out your requirements first. I once got into
>>>>>> same situation as your's and believe me reinventing the wheel (tailor made
>>>>>> solution) for exactly the requirement was the best thing. Atleast I know
>>>>>> I've created something which is doing all that I needed and the time spent
>>>>>> on this was less than a week !!
>>>>>>
>>>>>> Its upto you and your requirements to choose the solution.
>>>>>>
>>>>>> BR
>>>>>> Sammy Go.
>>>>>> On Aug 16, 2012 8:11 PM, "Abaci" <abaci64 at gmail.com> wrote:
>>>>>>
>>>>>>>  http://www.ictdialer.org/
>>>>>>>
>>>>>>> On 8/16/2012 9:53 AM, Obie Brown wrote:
>>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>>  I really want to work with an Open Source dialer and FreeSWITCH
>>>>>>> but the only thing I can find for this is Newfies Dialer which does not
>>>>>>> have features of a Auto Dialer as I need. Also its use for Voice
>>>>>>> Broadcasting is limited.
>>>>>>>
>>>>>>>  My question is. What other Auto Dialers that run on FreeSWITCH are
>>>>>>> there. If any.
>>>>>>>
>>>>>>>
>>>>>>>  Thanks,
>>>>>>> Obie Brown
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>>>>>
>>>>>>> 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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at endigotech.com  Thu Aug 23 16:25:53 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 23 Aug 2012 08:25:53 -0400
Subject: [Freeswitch-users] Voice broadcasting
In-Reply-To: <CAOL0t+LYhOBid83cWPwvUEPqyE309Vm6DC7ZkoQWO42CVUp4Og@mail.gmail.com>
References: <CAAdTMa-Qz1UyH+wXKEGXRBHF=Hjh7S_jpxe_BSphXgsrYpo_aw@mail.gmail.com>
	<CC359325.3A8AE%krice@freeswitch.org>
	<CAKV-jFzQizHQAbUEQxCPdi_Wc9odhcL0j7oc9grHX=kceZ3-aQ@mail.gmail.com>
	<CAAdTMa-wwTiyZ+=caecU8mC5K8Z2A_2Atnq4yonDRz-172BU5g@mail.gmail.com>
	<CAOL0t+LYhOBid83cWPwvUEPqyE309Vm6DC7ZkoQWO42CVUp4Og@mail.gmail.com>
Message-ID: <CAKV-jFykvFmKSPf1ZGL0tdth2ZJPyk2tX0J2aiWJHYW5Zg7vRA@mail.gmail.com>

Tahir,

Please be careful about marketing your product. This is the second email
within 24 hours from your company about ICTDialer.

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 23, 2012 3:21 AM, "tahir almas" <tahir at ictinnovations.com> wrote:

> I recommend to check ICTDialer that has simpler design and based on Plivo
> / Freeswitch http://www.ictdialer.org also supporting IVR Designer to
> draw your custom outbound IVR with drag and drop feature  for Interactive
> Voice Broadcasting , ICTDialer also support SMS Messaging and Fax
> broadcasting / blasting
>
> Regards
> *Tahir Almas*
>
> Managing Partner
> ICT Innovations
> http://www.ictinnovations.com
> Leveraging open source in ICT
>
>
> ****************************************************************************************************************
> NOTICE OF CONFIDENTIALITY
> This communication including any information transmitted with it is
> intended only for the use of the addressees and is confidential and may
> be protected by legal privilege . If you are not an intended recipient, be
> aware that any disclosure, copying, distribution or use of this e-mail or
> any attachment is prohibited. If you have received this e-mail in error,
> please notify us immediately by returning it to the sender and delete this
> copy from your system. Thank you for your cooperation.
>
>
>
>
> On Wed, Jul 25, 2012 at 11:50 PM, BookBag <asaad2 at gmail.com> wrote:
>
>> thanks everyone, I'll give it a try
>>
>>
>> On Wed, Jul 25, 2012 at 1:11 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>>
>>> Is newfies dialer a good fit for this case?
>>>
>>> Brian Foster
>>> Endigo Computer LLC
>>>
>>> Sent from a mobile device.
>>> On Jul 25, 2012 1:14 PM, "Ken Rice" <krice at freeswitch.org> wrote:
>>>
>>>>  You need a queue manager and all sorts of other instrumentation that
>>>> is outside the scope of FreeSWITCH itself.
>>>>
>>>> The short answer, yes you can do it, the not so short answer, theres a
>>>> lot of work to do to turn FreeSWITCH into a dialer.
>>>>
>>>> You might dig around in google to find some things that do robocalls on
>>>> freeswitch already...
>>>>
>>>>
>>>> On 7/25/12 11:56 AM, "BookBag" <asaad2 at gmail.com> wrote:
>>>>
>>>> thanx lol :-)
>>>>
>>>> But is there a way to have it call from a per-determined list
>>>>
>>>> On Wed, Jul 25, 2012 at 11:48 AM, Michael Collins <msc at freeswitch.org>
>>>> wrote:
>>>>
>>>> >From the fs_cli:
>>>> originate {ignore_early_media=true}sofia/gateway/your_gw_name/
>>>> 18005551212 <tel:18005551212>  &playback("/path/to/file.wav")
>>>>
>>>> Please don't abuse your new found power! :)
>>>> -MC
>>>>
>>>>
>>>> On Wed, Jul 25, 2012 at 9:39 AM, BookBag <asaad2 at gmail.com> wrote:
>>>>
>>>> Hello all, can anyone direct to some information on how I can setup a
>>>> voice broadcast on FS. I tried looking in the wiki but it justs redirects
>>>> you to heading "Robocalls", which in turn redirect you "voice broadcast"
>>>> but there is no information under it.
>>>>
>>>> Basically I just want to call about 100 people or so and play a
>>>> recording and then hangup.
>>>>
>>>>
>>>> If anyone could help me , i'd really appreciate it.
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> ------------------------------
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 vaad.fabi at gmail.com  Thu Aug 23 16:29:00 2012
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Thu, 23 Aug 2012 15:29:00 +0300
Subject: [Freeswitch-users] TLS documentation inaccuracy
In-Reply-To: <5033C5B1.3070509@hw.ac.uk>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>	<503226CE.7040307@hw.ac.uk>
	<50323D6A.3020708@anew.com.ve>	<2526639.vT3cT5qGKG@virtex>
	<5033C5B1.3070509@hw.ac.uk>
Message-ID: <5036220C.9080203@gmail.com>

  Hi all,

There are some TLS documentation inaccuracy 
(http://wiki.freeswitch.org/wiki/Tls):

1.  " ./gentls_cert create_server -cn pbx.freeswitch.org -alt 
DNS:pbx.freeswitch.org -org freeswitch.org"
There is no option "create_server" for gentls_cert , as i can see only 
setup,create,remove.

2.  "./gentls_cert create_client -cn Client1 -out Client1"
There is no option "create_client" for gentls_cert.
And there is no exactly information for softphone certificate 
generation. For Windows we need .crt file certificate, but how it could 
be generated (.pem only generated) ???

Please fix doc and gentls_cert script, or explain all procedures. Thx.

-- 
Best Regards,
Vadim F.



From david at styleflare.com  Thu Aug 23 16:37:12 2012
From: david at styleflare.com (David J)
Date: Thu, 23 Aug 2012 08:37:12 -0400
Subject: [Freeswitch-users] Voice broadcasting
In-Reply-To: <CAKV-jFykvFmKSPf1ZGL0tdth2ZJPyk2tX0J2aiWJHYW5Zg7vRA@mail.gmail.com>
References: <CAAdTMa-Qz1UyH+wXKEGXRBHF=Hjh7S_jpxe_BSphXgsrYpo_aw@mail.gmail.com>
	<CC359325.3A8AE%krice@freeswitch.org>
	<CAKV-jFzQizHQAbUEQxCPdi_Wc9odhcL0j7oc9grHX=kceZ3-aQ@mail.gmail.com>
	<CAAdTMa-wwTiyZ+=caecU8mC5K8Z2A_2Atnq4yonDRz-172BU5g@mail.gmail.com>
	<CAOL0t+LYhOBid83cWPwvUEPqyE309Vm6DC7ZkoQWO42CVUp4Og@mail.gmail.com>
	<CAKV-jFykvFmKSPf1ZGL0tdth2ZJPyk2tX0J2aiWJHYW5Zg7vRA@mail.gmail.com>
Message-ID: <CAPoPLj_QVqmA43PWM9rpzWrAii3k_3kp4BhYxngip+A84NbQAw@mail.gmail.com>

Brian.

I will agree with you only if ICT dialer is a commercial product.

Otherwise to mention it on list should be OK as long as its free switch
related.
On Aug 23, 2012 8:30 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> Tahir,
>
> Please be careful about marketing your product. This is the second email
> within 24 hours from your company about ICTDialer.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Aug 23, 2012 3:21 AM, "tahir almas" <tahir at ictinnovations.com> wrote:
>
>> I recommend to check ICTDialer that has simpler design and based on Plivo
>> / Freeswitch http://www.ictdialer.org also supporting IVR Designer to
>> draw your custom outbound IVR with drag and drop feature  for Interactive
>> Voice Broadcasting , ICTDialer also support SMS Messaging and Fax
>> broadcasting / blasting
>>
>> Regards
>> *Tahir Almas*
>>
>> Managing Partner
>> ICT Innovations
>> http://www.ictinnovations.com
>> Leveraging open source in ICT
>>
>>
>> ****************************************************************************************************************
>> NOTICE OF CONFIDENTIALITY
>> This communication including any information transmitted with it is
>> intended only for the use of the addressees and is confidential and may
>> be protected by legal privilege . If you are not an intended recipient, be
>> aware that any disclosure, copying, distribution or use of this e-mail or
>> any attachment is prohibited. If you have received this e-mail in error,
>> please notify us immediately by returning it to the sender and delete this
>> copy from your system. Thank you for your cooperation.
>>
>>
>>
>>
>> On Wed, Jul 25, 2012 at 11:50 PM, BookBag <asaad2 at gmail.com> wrote:
>>
>>> thanks everyone, I'll give it a try
>>>
>>>
>>> On Wed, Jul 25, 2012 at 1:11 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>>>
>>>> Is newfies dialer a good fit for this case?
>>>>
>>>> Brian Foster
>>>> Endigo Computer LLC
>>>>
>>>> Sent from a mobile device.
>>>> On Jul 25, 2012 1:14 PM, "Ken Rice" <krice at freeswitch.org> wrote:
>>>>
>>>>>  You need a queue manager and all sorts of other instrumentation that
>>>>> is outside the scope of FreeSWITCH itself.
>>>>>
>>>>> The short answer, yes you can do it, the not so short answer, theres a
>>>>> lot of work to do to turn FreeSWITCH into a dialer.
>>>>>
>>>>> You might dig around in google to find some things that do robocalls
>>>>> on freeswitch already...
>>>>>
>>>>>
>>>>> On 7/25/12 11:56 AM, "BookBag" <asaad2 at gmail.com> wrote:
>>>>>
>>>>> thanx lol :-)
>>>>>
>>>>> But is there a way to have it call from a per-determined list
>>>>>
>>>>> On Wed, Jul 25, 2012 at 11:48 AM, Michael Collins <msc at freeswitch.org>
>>>>> wrote:
>>>>>
>>>>> >From the fs_cli:
>>>>> originate {ignore_early_media=true}sofia/gateway/your_gw_name/
>>>>> 18005551212 <tel:18005551212>  &playback("/path/to/file.wav")
>>>>>
>>>>> Please don't abuse your new found power! :)
>>>>> -MC
>>>>>
>>>>>
>>>>> On Wed, Jul 25, 2012 at 9:39 AM, BookBag <asaad2 at gmail.com> wrote:
>>>>>
>>>>> Hello all, can anyone direct to some information on how I can setup a
>>>>> voice broadcast on FS. I tried looking in the wiki but it justs redirects
>>>>> you to heading "Robocalls", which in turn redirect you "voice broadcast"
>>>>> but there is no information under it.
>>>>>
>>>>> Basically I just want to call about 100 people or so and play a
>>>>> recording and then hangup.
>>>>>
>>>>>
>>>>> If anyone could help me , i'd really appreciate it.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> ------------------------------
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> Join Us At ClueCon - Aug 7-9, 2012
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> Join Us At ClueCon - Aug 7-9, 2012
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 alex at thewinelake.com  Thu Aug 23 16:38:23 2012
From: alex at thewinelake.com (Alex)
Date: Thu, 23 Aug 2012 13:38:23 +0100
Subject: [Freeswitch-users] ipv6 - any fans?
In-Reply-To: <5036220C.9080203@gmail.com>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>	<503226CE.7040307@hw.ac.uk>
	<50323D6A.3020708@anew.com.ve>	<2526639.vT3cT5qGKG@virtex>
	<5033C5B1.3070509@hw.ac.uk> <5036220C.9080203@gmail.com>
Message-ID: <5036243F.4060601@thewinelake.com>

We've been wondering about trying to offer a service on ipv6 as it would 
get around various LAN issues.

Just wondered what experiences (good or bad) people here have had with 
it and also why there isn't more "'buzz"?


From bdfoster at endigotech.com  Thu Aug 23 17:04:14 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 23 Aug 2012 09:04:14 -0400
Subject: [Freeswitch-users] ipv6 - any fans?
In-Reply-To: <5036243F.4060601@thewinelake.com>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in> <503226CE.7040307@hw.ac.uk>
	<50323D6A.3020708@anew.com.ve> <2526639.vT3cT5qGKG@virtex>
	<5033C5B1.3070509@hw.ac.uk> <5036220C.9080203@gmail.com>
	<5036243F.4060601@thewinelake.com>
Message-ID: <CAKV-jFy=GanJ1mu6TDVGm_HUwHRUJgcYbVQE0kh1nFO2CgRNPQ@mail.gmail.com>

Probably lack of cost-benefit for established companies. Why fix it if it
ain't broke? Also why invest in something that really isn't widely adopted
yet?

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 23, 2012 8:39 AM, "Alex" <alex at thewinelake.com> wrote:

> We've been wondering about trying to offer a service on ipv6 as it would
> get around various LAN issues.
>
> Just wondered what experiences (good or bad) people here have had with
> it and also why there isn't more "'buzz"?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 dgarcia at anew.com.ve  Thu Aug 23 17:12:33 2012
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Thu, 23 Aug 2012 08:42:33 -0430
Subject: [Freeswitch-users] FreeSWITCH Auto Dialer - Open source
In-Reply-To: <CA+aeUL8kFrwZR=worjfsozAv0Vs+jYLSDGFezvBVQhK3C8Gxcw@mail.gmail.com>
References: <CA+aeUL_D-1sgqxvxshU9oqELh66Ck-_kTxesMCCjnv3DgDu-yg@mail.gmail.com>
	<502D0CAC.8030905@gmail.com>
	<CAJUJwtg7vDr+MDbTvXJJk4HRAr38CTOYY3mRpWwBui0erBXLeg@mail.gmail.com>
	<CAOL0t+LwiTzxXY_VtA6FfN34NzQZYNcBHTHiFfrCK3ZyJMsvYw@mail.gmail.com>
	<CALNZuwnkQ1+nNHECNOM6hs8xCTV1wx3-ucevp01ur8bW5Km1tQ@mail.gmail.com>
	<CA+aeUL9LUZjmJp=LwG2NijHXcZvU3ib+1DTVh1MkPjQapt2Ksw@mail.gmail.com>
	<CAKV-jFxr0b8sJoEto1PBonLGuqQh5GM5GWZkv+5EuO=VRz0AMQ@mail.gmail.com>
	<CALNZuwnv9RwLS2kBwc9x5NMg9TWzvo496FrpXFnPk0nNAxn7Zg@mail.gmail.com>
	<CA+aeUL8kFrwZR=worjfsozAv0Vs+jYLSDGFezvBVQhK3C8Gxcw@mail.gmail.com>
Message-ID: <50362C41.8050507@anew.com.ve>

Hi,

It seems they have fixed the error. I could see the demo


On 8/23/2012 7:09 AM, Obie Brown wrote:
> Yes looks very simple error. When its back I will check demo.
>
> Thanks everyone for there suggestions.
>
>
> Thanks,
> Obie Brown
>
>
> On Thu, Aug 23, 2012 at 8:57 PM, Avi Marcus <avi at avimarcus.net 
> <mailto:avi at avimarcus.net>> wrote:
>
>     Brian Foster I'm not worried about it, I'd like to see the demo...
>     -Avi
>
>
>     On Thu, Aug 23, 2012 at 2:07 PM, Brian Foster
>     <bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>> wrote:
>
>         They might be having issues with their database. Simple error
>         nothing to worry about.
>
>         Brian Foster
>         Endigo Computer LLC
>
>         Sent from a mobile device.
>
>         On Aug 23, 2012 4:25 AM, "Obie Brown" <obiebrown at gmail.com
>         <mailto:obiebrown at gmail.com>> wrote:
>
>             Yes I was trying to look at demo the there day (few days
>             ago) and there was error. Still seems to be error.
>
>             /PDOException/: SQLSTATE[42S02]: Base table or view not
>             found: 1146 Table 'call4smile.semaphore' doesn't exist:
>             SELECT expire, value FROM {semaphore} WHERE name = :name;
>             Array ( [:name] => variable_init ) in
>             /lock_may_be_available()/ (line
>             /167/ of//usr/ictbilling/includes/lock.inc/).
>
>             If you can fix tahir is this something you can fix?
>
>
>             Thanks,
>             Obie Brown
>
>
>             On Thu, Aug 23, 2012 at 5:45 PM, Avi Marcus
>             <avi at avimarcus.net <mailto:avi at avimarcus.net>> wrote:
>
>                 The demo seems broken: http://demo.ictdialer.org/
>
>                 -Avi
>
>
>                 On Thu, Aug 23, 2012 at 10:14 AM, tahir almas
>                 <tahir at ictinnovations.com
>                 <mailto:tahir at ictinnovations.com>> wrote:
>
>                     I will recommend to check ICTDialer
>                     http://www.ictdialer.org
>                     <http://www.ictdialer.org/> that is based on Plivo
>                     Communication framework / Freeswitch and support
>                     for standard voice broadcasting as well as
>                     Interactive Voice broadcasting also you have
>                     support IVR Designer to design your IVR with drag
>                     and drop feature
>
>                     Regards
>
>                     *Tahir Almas*
>
>                     Managing Partner
>                     ICT Innovations
>                     http://www.ictinnovations.com
>                     <http://www.ictinnovations.com/>
>                     Leveraging open source in ICT
>
>
>
>
>
>                     On Thu, Aug 16, 2012 at 11:41 PM, SamyGo
>                     <govoiper at gmail.com <mailto:govoiper at gmail.com>>
>                     wrote:
>
>
>                         Hi,
>
>                         Well I'd say to figure out your requirements
>                         first. I once got into same situation as
>                         your's and believe me reinventing the wheel
>                         (tailor made solution) for exactly the
>                         requirement was the best thing. Atleast I know
>                         I've created something which is doing all that
>                         I needed and the time spent on this was less
>                         than a week !!
>
>                         Its upto you and your requirements to choose
>                         the solution.
>
>                         BR
>                         Sammy Go.
>
>                         On Aug 16, 2012 8:11 PM, "Abaci"
>                         <abaci64 at gmail.com <mailto:abaci64 at gmail.com>>
>                         wrote:
>
>                             http://www.ictdialer.org/
>
>                             On 8/16/2012 9:53 AM, Obie Brown wrote:
>>                             Hi,
>>
>>                             I really want to work with an Open Source
>>                             dialer and FreeSWITCH but the only thing
>>                             I can find for this is Newfies Dialer
>>                             which does not have features of a Auto
>>                             Dialer as I need. Also its use for Voice
>>                             Broadcasting is limited.
>>
>>                             My question is. What other Auto Dialers
>>                             that run on FreeSWITCH are there. If any.
>>
>>
>>                             Thanks,
>>                             Obie Brown
>>
>>
>>                             _________________________________________________________________________
>>                             Professional FreeSWITCH Consulting Services:
>>                             consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>                             http://www.freeswitchsolutions.com  <http://www.freeswitchsolutions.com/>
>>
>>                             
>>                               </>
>>
>>                             Official FreeSWITCH Sites
>>                             http://www.freeswitch.org  <http://www.freeswitch.org/>
>>                             http://wiki.freeswitch.org  <http://wiki.freeswitch.org/>
>>                             http://www.cluecon.com  <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  <http://www.freeswitch.org/>
>
>
>                             _________________________________________________________________________
>                             Professional FreeSWITCH Consulting Services:
>                             consulting at freeswitch.org
>                             <mailto:consulting at freeswitch.org>
>                             http://www.freeswitchsolutions.com
>                             <http://www.freeswitchsolutions.com/>
>
>                             FreeSWITCH-powered IP PBX: The CudaTel
>                             Communication Server
>                             
>                             </>
>
>                             Official FreeSWITCH Sites
>                             http://www.freeswitch.org
>                             <http://www.freeswitch.org/>
>                             http://wiki.freeswitch.org
>                             <http://wiki.freeswitch.org/>
>                             http://www.cluecon.com
>                             <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
>                             <http://www.freeswitch.org/>
>
>
>                         _________________________________________________________________________
>                         Professional FreeSWITCH Consulting Services:
>                         consulting at freeswitch.org
>                         <mailto:consulting at freeswitch.org>
>                         http://www.freeswitchsolutions.com
>                         <http://www.freeswitchsolutions.com/>
>
>                         FreeSWITCH-powered IP PBX: The CudaTel
>                         Communication Server
>                          </>
>
>                         Official FreeSWITCH Sites
>                         http://www.freeswitch.org
>                         <http://www.freeswitch.org/>
>                         http://wiki.freeswitch.org
>                         <http://wiki.freeswitch.org/>
>                         http://www.cluecon.com <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
>                         <http://www.freeswitch.org/>
>
>
>
>                     _________________________________________________________________________
>                     Professional FreeSWITCH Consulting Services:
>                     consulting at freeswitch.org
>                     <mailto:consulting at freeswitch.org>
>                     http://www.freeswitchsolutions.com
>                     <http://www.freeswitchsolutions.com/>
>
>                     FreeSWITCH-powered IP PBX: The CudaTel
>                     Communication Server
>                      </>
>
>                     Official FreeSWITCH Sites
>                     http://www.freeswitch.org <http://www.freeswitch.org/>
>                     http://wiki.freeswitch.org
>                     <http://wiki.freeswitch.org/>
>                     http://www.cluecon.com <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 <http://www.freeswitch.org/>
>
>
>
>                 _________________________________________________________________________
>                 Professional FreeSWITCH Consulting Services:
>                 consulting at freeswitch.org
>                 <mailto:consulting at freeswitch.org>
>                 http://www.freeswitchsolutions.com
>                 <http://www.freeswitchsolutions.com/>
>
>                 FreeSWITCH-powered IP PBX: The CudaTel Communication
>                 Server
>                  </>
>
>                 Official FreeSWITCH Sites
>                 http://www.freeswitch.org <http://www.freeswitch.org/>
>                 http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>                 http://www.cluecon.com <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 <http://www.freeswitch.org/>
>
>
>
>             _________________________________________________________________________
>             Professional FreeSWITCH Consulting Services:
>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>             http://www.freeswitchsolutions.com
>             <http://www.freeswitchsolutions.com/>
>
>             
>              </>
>
>             Official FreeSWITCH Sites
>             http://www.freeswitch.org <http://www.freeswitch.org/>
>             http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>             http://www.cluecon.com <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 <http://www.freeswitch.org/>
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>         <http://www.freeswitchsolutions.com/>
>
>         
>          </>
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org <http://www.freeswitch.org/>
>         http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>         http://www.cluecon.com <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 <http://www.freeswitch.org/>
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com/>
>
>     
>      </>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org <http://www.freeswitch.org/>
>     http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>     http://www.cluecon.com <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 <http://www.freeswitch.org/>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2197 / Virus Database: 2437/5219 - Release Date: 08/23/12
>


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

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Tel?fono: +58 212 9081842
Cel: +58 412 2221515
dgarcia at anew.com.ve
http://www.anew.com.ve
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From eranga.erl at gmail.com  Thu Aug 23 16:02:58 2012
From: eranga.erl at gmail.com (Eranga Udesh)
Date: Thu, 23 Aug 2012 17:32:58 +0530
Subject: [Freeswitch-users] Mod Erlang Event in Freeswitch
Message-ID: <CALg+e6z0=xhmohAJHEoapsP4oGgBzg1uyQgBe0NDJSeRrEa9RA@mail.gmail.com>

Hi,

I'm trying mod_erlang_event module in Freeswitch. However when a call comes
to extension 111, the Erlang node receives RPC call correctly (my config as
below) but it seems Freeswitch module does not receive the returned {Ref,
NewPid} tuple of the Erlang function. I use Erlang R15B01 version.

I also tried with Registered process configuration as in the Mod Erlang
Event wiki, but there also it timeout. I wonder if it is an incompatibility
with Erlang 15B?

Any ideas what could be wrong?

Config:

<extension name="to_erlang" continue="true">
   <condition field="destination_number" expression="^111$" >
       <action application="erlang" data="fsivr:new_call ivr at linux" />
   </condition>
</extension>

Output in freeswitch debug:

2012-08-23 17:06:34.057997 [DEBUG] mod_erlang_event.c:1519 enter
erlang_outbound_function fsivr:new_call ivr at linux
2012-08-23 17:06:34.057997 [DEBUG] mod_erlang_event.c:1525 Creating new
listener for session
2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1527 Launching new
listener
2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1537 Creating new
spawned session for listener
2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1423 rpc call:
fsivr:new_call(Ref)
2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1008 Connection Open
2012-08-23 17:06:39.177997 [WARNING] mod_erlang_event.c:1436 Timed out when
waiting for outbound pid 12.0.0 at freeswitch@linux
404fc3c7-45d1-4194-ad37-1e118fdee8d0
2012-08-23 17:06:39.177997 [DEBUG] switch_channel.c:2919
(sofia/internal/5000 at linux4) Callstate Change RINGING -> HANGUP

Thanks in advance.
- Eranga
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From mgg at giagnocavo.net  Thu Aug 23 17:57:24 2012
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Thu, 23 Aug 2012 13:57:24 +0000
Subject: [Freeswitch-users] ipv6 - any fans?
In-Reply-To: <CAKV-jFy=GanJ1mu6TDVGm_HUwHRUJgcYbVQE0kh1nFO2CgRNPQ@mail.gmail.com>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<503226CE.7040307@hw.ac.uk>	<50323D6A.3020708@anew.com.ve>
	<2526639.vT3cT5qGKG@virtex>	<5033C5B1.3070509@hw.ac.uk>
	<5036220C.9080203@gmail.com>	<5036243F.4060601@thewinelake.com>
	<CAKV-jFy=GanJ1mu6TDVGm_HUwHRUJgcYbVQE0kh1nFO2CgRNPQ@mail.gmail.com>
Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FB0437@BLUPRD0711MB413.namprd07.prod.outlook.com>

Yea, until IPv4 is under such a crunch that it's _actually_ hard to get IPs, IPv6 adoption won't have that much pressure. It'd probably be cool to offer IPv6 VoIP services, but I wouldn't expect much actual usage for a while.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster
Sent: Thursday, August 23, 2012 7:04 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] ipv6 - any fans?


Probably lack of cost-benefit for established companies. Why fix it if it ain't broke? Also why invest in something that really isn't widely adopted yet?

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 23, 2012 8:39 AM, "Alex" <alex at thewinelake.com<mailto:alex at thewinelake.com>> wrote:
We've been wondering about trying to offer a service on ipv6 as it would
get around various LAN issues.

Just wondered what experiences (good or bad) people here have had with
it and also why there isn't more "'buzz"?

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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From mgg at giagnocavo.net  Thu Aug 23 17:57:25 2012
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Thu, 23 Aug 2012 13:57:25 +0000
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <CD9593DE-1F02-4135-AEDD-7ED7C6A5FB22@visionutveckling.se>
References: <5034019b.c456b60a.3fd2.ffffdc5cSMTPIN_ADDED@mx.google.com>
	<CALNZuwmN+xfGkW0HqqDpXdJ4WaW5o59BcX-RiuKdNo18QH5D7g@mail.gmail.com>,
	<004201cd7fed$b22ef150$168cd3f0$@caulfield@intermetro.net>
	<CD9593DE-1F02-4135-AEDD-7ED7C6A5FB22@visionutveckling.se>
Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FB043F@BLUPRD0711MB413.namprd07.prod.outlook.com>

Just playing along with the whole "we can't get real hardware" idea:

 - Try a 32-bit build, this may decrease memory usage significantly.
 - You didn't mention how the memory is actually used (is that 85% physically mapped?). Each thread gets a 240K (I think) allocation by default. You might try reducing that, but you'll need to test it and make sure you don't start overflowing.

As much as I disagree with the "ten-thousand-threads" model, FS isn't going to change it anytime soon (and hey, it's working). And, I am at a loss as to what business model has so many calls, and can't afford a $500 server. So, just buy a server and move on.

-Michael

-------
It is not so much the cps but the concurrent calls.

I am looking at maintaining a 30 cps with a 5 min. avg. call length which results in a sustained 9000 concurrent calls.

This results in 85% mem. usage on a 4 GB machine, However CPU is at 25% (total) on a dual core with hyper threading.

Current hardware does not support increase memory.

I am striving for about 75% cpu (appox. 90 cps).

Jason


From: 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 Avi Marcus
Sent: Tuesday, August 21, 2012 2:58 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSWITCH Question

Do you have such a large cps/concurrent volume that you're actually seeing a performance hit?

-Avi

On Wed, Aug 22, 2012 at 12:35 AM, Jason Caulfield <jason.caulfield at intermetro.net<mailto:jason.caulfield at intermetro.net>> wrote:
I am using FreeSWITCH in media bypass mode to decrease memory and cpu usage.

I notice that mod_sofia creates threads for each call leg to maintain session context.

Do you know of a way to configure FreeSWITCH to use a table to maintain the context when in media bypass mode to reduce the number of threads?

I am hoping that this will speed things up by reducing thread context switching and reduce memory usage by decreasing memory allocation for each thread.

Thanks for the help,
Jason

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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:50340a7d32766392456629!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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http://www.freeswitch.org


!DSPAM:50340a7d32766392456629!

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 cmason at frontiernetworks.ca  Thu Aug 23 18:33:06 2012
From: cmason at frontiernetworks.ca (Colin Mason)
Date: Thu, 23 Aug 2012 10:33:06 -0400
Subject: [Freeswitch-users] Shared Call Appearance and Transferring Calls on
	Cisco SPA504G
Message-ID: <0D1C698866F66045A6201FD0F59CAC90014678A921@EX.frontier.local>

So I have been experimenting with a key system for an office with 3 Cisco SPA504G phones and FreeSWITCH 1.2.1. They require 3 shared lines. I have inbound and outbound calls working properly with the shared lines and the phones are configured properly with shared lines, Broadcom etc. Visually everything seems to work with the line notifications on the 3 phones.

My internal profile has:

    <param name="manage-shared-appearance" value="true"/>
    <param name="manage-presence" value="true"/>
    <param name="dbname" value="share_presence"/>
    <param name="presence-hosts" value="$${domain}"/>
    <param name="multiple-registrations" value="true"/>
    <param name="force-register-domain" value="$${domain}"/>
    <param name="force-subscription-domain" value="$${domain}"/>
    <param name="force-register-db-domain" value="$${domain}"/>
    <param name="presence-probe-on-register" value="true"/>
    <param name="presence-privacy" value="$${presence_privacy}"/>
    <param name="send-presence-on-register" value="true"/>

I am having problems with transferring calls. If I put a call on hold on phone 1 and press the line 1 button on phone 1, the call resumes just fine. But if I put a call on hold on phone 1 and try to resume the call on phone 2 by pressing the blinking red line 1 button, the phone tries to establish a new call to the user associated with line 1 instead of taking (transferring) the call from phone 1 to phone 2.

I was wondering if anybody could help?

Colin Mason
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From kris at kriskinc.com  Thu Aug 23 18:42:12 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 23 Aug 2012 10:42:12 -0400
Subject: [Freeswitch-users] ipv6 - any fans?
In-Reply-To: <5036243F.4060601@thewinelake.com>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in> <503226CE.7040307@hw.ac.uk>
	<50323D6A.3020708@anew.com.ve> <2526639.vT3cT5qGKG@virtex>
	<5033C5B1.3070509@hw.ac.uk> <5036220C.9080203@gmail.com>
	<5036243F.4060601@thewinelake.com>
Message-ID: <CAKDfjgdVYJBS0_WVdsSHE7=PPwRYW_RCSb0Pdvx0Yfbn4qWy4A@mail.gmail.com>

Assuming you're in the US (or most other parts of the world besides
Asia); it won't work.  Here's why:

Almost no last mile providers will deliver native IPv6 (currently).
Your customers would have to use a tunneling service like HE to even
get to you.  First there are hassles setting up the tunnel in the
first place, then there are issues with latency, packet loss, jitter,
etc using a tunnel service that has to haul all of your IPv6 in IPv4
traffic back to their gateway in a central location.  VoIP is just
about the worst service you could use via an IPv6 tunnel.

You could try to offer your own IPv6 tunneling service but there's
still the issue of customers setting up there end of the tunnel.
Besides, encapsulation sucks.

The IPv4 global routing table has at least 424,000 networks:

http://bgp.potaroo.net/index-bgp.html

The IPv6 global routing table has 10,215 networks:

http://bgp.potaroo.net/v6/as2.0/

Granted IPv6 networks tend to be better organized and more "compact"
(at least as far as advertisements are concerned) but that's still a
factor of 40:1.

On Thu, Aug 23, 2012 at 8:38 AM, Alex <alex at thewinelake.com> wrote:
> We've been wondering about trying to offer a service on ipv6 as it would
> get around various LAN issues.
>
> Just wondered what experiences (good or bad) people here have had with
> it and also why there isn't more "'buzz"?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Kristian Kielhofner


From kris at kriskinc.com  Thu Aug 23 18:43:25 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 23 Aug 2012 10:43:25 -0400
Subject: [Freeswitch-users] ipv6 - any fans?
In-Reply-To: <CAKDfjgdVYJBS0_WVdsSHE7=PPwRYW_RCSb0Pdvx0Yfbn4qWy4A@mail.gmail.com>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in> <503226CE.7040307@hw.ac.uk>
	<50323D6A.3020708@anew.com.ve> <2526639.vT3cT5qGKG@virtex>
	<5033C5B1.3070509@hw.ac.uk> <5036220C.9080203@gmail.com>
	<5036243F.4060601@thewinelake.com>
	<CAKDfjgdVYJBS0_WVdsSHE7=PPwRYW_RCSb0Pdvx0Yfbn4qWy4A@mail.gmail.com>
Message-ID: <CAKDfjgeY0UGZqm33Zrb=XdhEcgb9paz7sJ+dmi8_1PYZi35RaA@mail.gmail.com>

Sorry, I just woke up: "there end of the tunnel" = "their end of the tunnel".

On Thu, Aug 23, 2012 at 10:42 AM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> Assuming you're in the US (or most other parts of the world besides
> Asia); it won't work.  Here's why:
>
> Almost no last mile providers will deliver native IPv6 (currently).
> Your customers would have to use a tunneling service like HE to even
> get to you.  First there are hassles setting up the tunnel in the
> first place, then there are issues with latency, packet loss, jitter,
> etc using a tunnel service that has to haul all of your IPv6 in IPv4
> traffic back to their gateway in a central location.  VoIP is just
> about the worst service you could use via an IPv6 tunnel.
>
> You could try to offer your own IPv6 tunneling service but there's
> still the issue of customers setting up there end of the tunnel.
> Besides, encapsulation sucks.
>
> The IPv4 global routing table has at least 424,000 networks:
>
> http://bgp.potaroo.net/index-bgp.html
>
> The IPv6 global routing table has 10,215 networks:
>
> http://bgp.potaroo.net/v6/as2.0/
>
> Granted IPv6 networks tend to be better organized and more "compact"
> (at least as far as advertisements are concerned) but that's still a
> factor of 40:1.
>
> On Thu, Aug 23, 2012 at 8:38 AM, Alex <alex at thewinelake.com> wrote:
>> We've been wondering about trying to offer a service on ipv6 as it would
>> get around various LAN issues.
>>
>> Just wondered what experiences (good or bad) people here have had with
>> it and also why there isn't more "'buzz"?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Kristian Kielhofner



-- 
Kristian Kielhofner


From kris at kriskinc.com  Thu Aug 23 19:02:22 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 23 Aug 2012 11:02:22 -0400
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FB043F@BLUPRD0711MB413.namprd07.prod.outlook.com>
References: <5034019b.c456b60a.3fd2.ffffdc5cSMTPIN_ADDED@mx.google.com>
	<CALNZuwmN+xfGkW0HqqDpXdJ4WaW5o59BcX-RiuKdNo18QH5D7g@mail.gmail.com>
	<CD9593DE-1F02-4135-AEDD-7ED7C6A5FB22@visionutveckling.se>
	<63B00DD1DA6A364E9F64A3A0BD2FE7B612FB043F@BLUPRD0711MB413.namprd07.prod.outlook.com>
Message-ID: <CAKDfjgcnR4mTNfuqWckmsLOL_B6-p9fq8DzbiWpvSDgcCH4YEA@mail.gmail.com>

"And, I am at a loss as to what business model has so many calls, and
can't afford a $500 server. So, just buy a server and move on."

Once again, +1.

On Thu, Aug 23, 2012 at 9:57 AM, Michael Giagnocavo <mgg at giagnocavo.net> wrote:
> Just playing along with the whole "we can't get real hardware" idea:
>
>  - Try a 32-bit build, this may decrease memory usage significantly.
>  - You didn't mention how the memory is actually used (is that 85% physically mapped?). Each thread gets a 240K (I think) allocation by default. You might try reducing that, but you'll need to test it and make sure you don't start overflowing.
>
> As much as I disagree with the "ten-thousand-threads" model, FS isn't going to change it anytime soon (and hey, it's working). And, I am at a loss as to what business model has so many calls, and can't afford a $500 server. So, just buy a server and move on.
>
> -Michael
>
> -------
> It is not so much the cps but the concurrent calls.
>
> I am looking at maintaining a 30 cps with a 5 min. avg. call length which results in a sustained 9000 concurrent calls.
>
> This results in 85% mem. usage on a 4 GB machine, However CPU is at 25% (total) on a dual core with hyper threading.
>
> Current hardware does not support increase memory.
>
> I am striving for about 75% cpu (appox. 90 cps).
>
> Jason
>
>
> From: 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 Avi Marcus
> Sent: Tuesday, August 21, 2012 2:58 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FreeSWITCH Question
>
> Do you have such a large cps/concurrent volume that you're actually seeing a performance hit?
>
> -Avi
>
> On Wed, Aug 22, 2012 at 12:35 AM, Jason Caulfield <jason.caulfield at intermetro.net<mailto:jason.caulfield at intermetro.net>> wrote:
> I am using FreeSWITCH in media bypass mode to decrease memory and cpu usage.
>
> I notice that mod_sofia creates threads for each call leg to maintain session context.
>
> Do you know of a way to configure FreeSWITCH to use a table to maintain the context when in media bypass mode to reduce the number of threads?
>
> I am hoping that this will speed things up by reducing thread context switching and reduce memory usage by decreasing memory allocation for each thread.
>
> Thanks for the help,
> Jason
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
>  
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> 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:50340a7d32766392456629!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
>  
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> 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:50340a7d32766392456629!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Kristian Kielhofner


From benkokakao at gmail.com  Thu Aug 23 19:04:03 2012
From: benkokakao at gmail.com (Christian Benke)
Date: Thu, 23 Aug 2012 17:04:03 +0200
Subject: [Freeswitch-users] RTP Timestamp changes after REFER - Question on
	RFC conformity
Message-ID: <CAAMQ8bRGeZm-7NWbRg4yoRZBqYTHEkXUCKUJJYgsEGZPZYNVgQ@mail.gmail.com>

Hi!

I had a problem with VOP Softclient(www.voiceoperatorpanel.com)  where
the callee had one-way-audio after a call was REFERred to him. After
lots of tracing, debugging and hairpulling i realized the problem lies
in the changing rtp-timestamp when the RTP-stream is switched from the
middleman to the initial call after the REFER. The Softclient was not
able to cope with the changed timestamp and ignored the incoming
RTP-packets, leading to no audio for the callee.

I was eventually able to solve the problem by activating
rtp-rewrite-timestamps on the profile(Also added it to
http://wiki.freeswitch.org/wiki/RTP_Issues#Voiceoperatorpanel_VOP).

However, i would like to know if FreeSWITCH/Sofia is working according
to the RFC and if the Softclient is to blame for the problem(So i can
file a bugreport with them).

In this thread, Brian West states that  it's ok to skip forward in
timestamps as long as the marker-bit is set:
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-July/060333.html
However, the Marker-bit is not set by FreeSWITCH when the REFER occurs.

I didn't find this stated in http://tools.ietf.org/html/rfc3550(But
"timestamp" is mentioned a lot, so i may have missed it) but there's
this bug-report for Asterisk, where the exact same problem is
described and eventually handled:
https://issues.asterisk.org/view.php?id=17007

Could someone with more insight please elaborate?

Best regards,
Christian


From leon at scarlet-internet.nl  Thu Aug 23 19:08:25 2012
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Thu, 23 Aug 2012 17:08:25 +0200
Subject: [Freeswitch-users] T.38 three re-invites then bye
Message-ID: <55B426C9-D9F2-41BC-90FB-DB3EC111C11B@scarlet-internet.nl>

Hi all,

I've been having a problem with a new provider with sending them faxes over T.38. I don't think FreeSWITCH is to blame at all, but maybe FS can fix my issue, or someone here can help me find the cause of the problem ?

The call flow is as follows:

CPE
1.1.1.1
     |
    V
2.2.2.2
FreeSWITCH
3.3.3.3
     |
    V
4.4.4.4
Provider

In FreeSWITCH, I've set t38-passthru to true on the profile where the CPE INVITEs to. Further, I'm doing late negotiation (while offering the a-leg codecs to the b-leg and the other way around). RTP is flowing through FS.

In short, this is happening:

CPE -> FS -> Provider		INVITE (with audio PCMA, PCMU codecs)
Provider -> FS -> CPE		200 OK (with audio PCMA)

CPE -> FS -> Provider		reINVITE (with T.38 in SDP)
Provider -> FS -> CPE		200 OK (with T.38 in SDP - although with different error correction, is this a problem?)

Provider -> FS -> CPE		reINVITE (again with T.38 in SDP)
CPE -> FS -> Provider		200 OK (with T.38 in SDP)

CPE -> FS -> Provider		reINVITE (with T.38 in SDP)
Provider -> FS -> CPE		BYE (of course, or it'll loop forever)

Does a setting in FreeSWITCH exist so that it can change the error correction (because both endpoints seem to want a different one ?) Could that be the cause of the problem ?

Hope anyone can help.

Thanks,

Leon





Here's the trace - I left out the tryings, acks, removed some unnecessary sip headers to make things a bit less verbose:


U 2012/08/23 15:34:16.334938 3.3.3.3:5060 -> 4.4.4.4:5060
INVITE sip:+888 at 4.4.4.4 SIP/2.0.
From: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
To: <sip:+888 at 4.4.4.4>.
Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
CSeq: 32528364 INVITE.
Contact: <sip:gw+provider-voip-tel at 3.3.3.3:5060;transport=udp;gw=provider-voip-tel>.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 253.
X-Designated-Service: fax/t38.
.
v=0.
o=FreeSWITCH 1345708964 1345708965 IN IP4 3.3.3.3.
s=FreeSWITCH.
c=IN IP4 3.3.3.3.
t=0 0.
m=audio 19892 RTP/AVP 8 0 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:30.


U 2012/08/23 15:34:18.192303 4.4.4.4:5060 -> 3.3.3.3:5060
SIP/2.0 180 Ringing.
From: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
To: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
CSeq: 32528364 INVITE.
Server: DC-SIP/2.0.
Content-Length: 0.
Contact: <sip:+888 at 4.4.4.4:5060;transport=udp>.
.


U 2012/08/23 15:34:23.904745 4.4.4.4:5060 -> 3.3.3.3:5060
SIP/2.0 200 OK.
From: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
To: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
CSeq: 32528364 INVITE.
Server: DC-SIP/2.0.
Contact: <sip:+888 at 4.4.4.4:5060;transport=udp>.
Content-Type: application/sdp.
Content-Length: 229.
.
v=0.
o=- 3554717663 3554717663 IN IP4 4.4.4.4.
s=-.
c=IN IP4 4.4.4.4.
t=0 0.
m=audio 22054 RTP/AVP 8 101.
a=ptime:30.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=silenceSupp:off - - - -.


U 2012/08/23 15:34:23.996849 3.3.3.3:5060 -> 4.4.4.4:5060
INVITE sip:+888 at 4.4.4.4:5060;transport=udp SIP/2.0.
From: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
To: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
CSeq: 32528365 INVITE.
Contact: <sip:gw+provider-voip-tel at 3.3.3.3:5060;transport=udp;gw=provider-voip-tel>.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 339.
X-Designated-Service: fax/t38.
.
v=0.
o=FreeSWITCH 1345708964 1345708966 IN IP4 3.3.3.3.
s=FreeSWITCH.
c=IN IP4 3.3.3.3.
t=0 0.
m=image 19892 udptl t38.
a=T38FaxVersion:0.
a=T38MaxBitRate:14400.
a=T38FaxTranscodingMMR.
a=T38FaxTranscodingJBIG.
a=T38FaxRateManagement:transferredTCF.
a=T38FaxMaxBuffer:500.
a=T38FaxMaxDatagram:512.
a=T38FaxUdpEC:t38UDPFEC.


U 2012/08/23 15:34:24.093598 4.4.4.4:5060 -> 3.3.3.3:5060
SIP/2.0 200 OK.
From: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
To: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
CSeq: 32528365 INVITE.
Server: DC-SIP/2.0.
Contact: <sip:+888 at 4.4.4.4:5060;transport=udp>.
Content-Type: application/sdp.
Content-Length: 272.
.
v=0.
o=- 3554717663 3554717664 IN IP4 4.4.4.4.
s=-.
c=IN IP4 4.4.4.4.
t=0 0.
m=image 22054 udptl t38.
a=T38FaxVersion:0.
a=T38MaxBitRate:14400.
a=T38FaxRateManagement:transferredTCF.
a=T38FaxMaxBuffer:72.
a=T38FaxMaxDatagram:316.
a=T38FaxUdpEC:t38UDPRedundancy.


U 2012/08/23 15:34:33.885255 4.4.4.4:5060 -> 3.3.3.3:5060
INVITE sip:gw+provider-voip-tel at 3.3.3.3:5060;transport=udp;gw=provider-voip-tel SIP/2.0.
Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
From: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
To: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
CSeq: 693734809 INVITE.
Contact: <sip:+888 at 4.4.4.4:5060;transport=udp>.
Content-Type: application/sdp.
Content-Length: 272.
.
v=0.
o=- 3554717663 3554717665 IN IP4 4.4.4.4.
s=-.
c=IN IP4 4.4.4.4.
t=0 0.
m=image 22054 udptl t38.
a=T38FaxVersion:0.
a=T38MaxBitRate:14400.
a=T38FaxRateManagement:transferredTCF.
a=T38FaxMaxBuffer:72.
a=T38FaxMaxDatagram:316.
a=T38FaxUdpEC:t38UDPRedundancy.


U 2012/08/23 15:34:33.939092 3.3.3.3:5060 -> 4.4.4.4:5060
SIP/2.0 200 OK.
From: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
To: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
CSeq: 693734809 INVITE.
Contact: <sip:gw+provider-voip-tel at 3.3.3.3:5060;transport=udp;gw=provider-voip-tel>.
Accept: application/sdp.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 345.
.
v=0.
o=FreeSWITCH 1345708964 1345708967 IN IP4 3.3.3.3.
s=FreeSWITCH.
c=IN IP4 3.3.3.3.
t=0 0.
m=image 19892 udptl t38.
a=T38FaxVersion:0.
a=T38MaxBitRate:14400.
a=T38FaxTranscodingMMR.
a=T38FaxTranscodingJBIG.
a=T38FaxRateManagement:transferredTCF.
a=T38FaxMaxBuffer:72.
a=T38FaxMaxDatagram:512.
a=T38FaxUdpEC:t38UDPRedundancy.


U 2012/08/23 15:34:33.950201 3.3.3.3:5060 -> 4.4.4.4:5060
INVITE sip:+888 at 4.4.4.4:5060;transport=udp SIP/2.0.
From: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
To: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
CSeq: 32528366 INVITE.
Contact: <sip:gw+provider-voip-tel at 3.3.3.3:5060;transport=udp;gw=provider-voip-tel>.
User-Agent: c4-provider-ims-intx.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 338.
X-Designated-Service: fax/t38.
.
v=0.
o=FreeSWITCH 1345708964 1345708968 IN IP4 3.3.3.3.
s=FreeSWITCH.
c=IN IP4 3.3.3.3.
t=0 0.
m=image 19892 udptl t38.
a=T38FaxVersion:0.
a=T38MaxBitRate:14400.
a=T38FaxTranscodingMMR.
a=T38FaxTranscodingJBIG.
a=T38FaxRateManagement:transferredTCF.
a=T38FaxMaxBuffer:72.
a=T38FaxMaxDatagram:512.
a=T38FaxUdpEC:t38UDPFEC.


U 2012/08/23 15:34:34.043165 4.4.4.4:5060 -> 3.3.3.3:5060
BYE sip:gw+provider-voip-tel at 3.3.3.3:5060;transport=udp;gw=provider-voip-tel SIP/2.0.
Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
From: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
To: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
CSeq: 693734810 BYE.
Content-Length: 0.
Reason: Q.850; cause=41;text="Temporary failure".
.




From anthony.minessale at gmail.com  Thu Aug 23 19:15:07 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 23 Aug 2012 10:15:07 -0500
Subject: [Freeswitch-users] T.38 three re-invites then bye
In-Reply-To: <55B426C9-D9F2-41BC-90FB-DB3EC111C11B@scarlet-internet.nl>
References: <55B426C9-D9F2-41BC-90FB-DB3EC111C11B@scarlet-internet.nl>
Message-ID: <CAKbxfG8q+wYnNV89UTGHfs3T5RQkR2yNxceunJiVTYusU2xATQ@mail.gmail.com>

You could hack the sdp in your dialplan by getting the variable and
re-setting it but I am not sure if that would work or not.
Otherwise you could try a proxy.

On Thu, Aug 23, 2012 at 10:08 AM, Leon de Rooij
<leon at scarlet-internet.nl> wrote:
> Hi all,
>
> I've been having a problem with a new provider with sending them faxes over T.38. I don't think FreeSWITCH is to blame at all, but maybe FS can fix my issue, or someone here can help me find the cause of the problem ?
>
> The call flow is as follows:
>
> CPE
> 1.1.1.1
>      |
>     V
> 2.2.2.2
> FreeSWITCH
> 3.3.3.3
>      |
>     V
> 4.4.4.4
> Provider
>
> In FreeSWITCH, I've set t38-passthru to true on the profile where the CPE INVITEs to. Further, I'm doing late negotiation (while offering the a-leg codecs to the b-leg and the other way around). RTP is flowing through FS.
>
> In short, this is happening:
>
> CPE -> FS -> Provider           INVITE (with audio PCMA, PCMU codecs)
> Provider -> FS -> CPE           200 OK (with audio PCMA)
>
> CPE -> FS -> Provider           reINVITE (with T.38 in SDP)
> Provider -> FS -> CPE           200 OK (with T.38 in SDP - although with different error correction, is this a problem?)
>
> Provider -> FS -> CPE           reINVITE (again with T.38 in SDP)
> CPE -> FS -> Provider           200 OK (with T.38 in SDP)
>
> CPE -> FS -> Provider           reINVITE (with T.38 in SDP)
> Provider -> FS -> CPE           BYE (of course, or it'll loop forever)
>
> Does a setting in FreeSWITCH exist so that it can change the error correction (because both endpoints seem to want a different one ?) Could that be the cause of the problem ?
>
> Hope anyone can help.
>
> Thanks,
>
> Leon
>
>
>
>
>
> Here's the trace - I left out the tryings, acks, removed some unnecessary sip headers to make things a bit less verbose:
>
>
> U 2012/08/23 15:34:16.334938 3.3.3.3:5060 -> 4.4.4.4:5060
> INVITE sip:+888 at 4.4.4.4 SIP/2.0.
> From: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
> To: <sip:+888 at 4.4.4.4>.
> Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
> CSeq: 32528364 INVITE.
> Contact: <sip:gw+provider-voip-tel at 3.3.3.3:5060;transport=udp;gw=provider-voip-tel>.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 253.
> X-Designated-Service: fax/t38.
> .
> v=0.
> o=FreeSWITCH 1345708964 1345708965 IN IP4 3.3.3.3.
> s=FreeSWITCH.
> c=IN IP4 3.3.3.3.
> t=0 0.
> m=audio 19892 RTP/AVP 8 0 101 13.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:30.
>
>
> U 2012/08/23 15:34:18.192303 4.4.4.4:5060 -> 3.3.3.3:5060
> SIP/2.0 180 Ringing.
> From: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
> To: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
> Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
> CSeq: 32528364 INVITE.
> Server: DC-SIP/2.0.
> Content-Length: 0.
> Contact: <sip:+888 at 4.4.4.4:5060;transport=udp>.
> .
>
>
> U 2012/08/23 15:34:23.904745 4.4.4.4:5060 -> 3.3.3.3:5060
> SIP/2.0 200 OK.
> From: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
> To: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
> Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
> CSeq: 32528364 INVITE.
> Server: DC-SIP/2.0.
> Contact: <sip:+888 at 4.4.4.4:5060;transport=udp>.
> Content-Type: application/sdp.
> Content-Length: 229.
> .
> v=0.
> o=- 3554717663 3554717663 IN IP4 4.4.4.4.
> s=-.
> c=IN IP4 4.4.4.4.
> t=0 0.
> m=audio 22054 RTP/AVP 8 101.
> a=ptime:30.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=silenceSupp:off - - - -.
>
>
> U 2012/08/23 15:34:23.996849 3.3.3.3:5060 -> 4.4.4.4:5060
> INVITE sip:+888 at 4.4.4.4:5060;transport=udp SIP/2.0.
> From: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
> To: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
> Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
> CSeq: 32528365 INVITE.
> Contact: <sip:gw+provider-voip-tel at 3.3.3.3:5060;transport=udp;gw=provider-voip-tel>.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 339.
> X-Designated-Service: fax/t38.
> .
> v=0.
> o=FreeSWITCH 1345708964 1345708966 IN IP4 3.3.3.3.
> s=FreeSWITCH.
> c=IN IP4 3.3.3.3.
> t=0 0.
> m=image 19892 udptl t38.
> a=T38FaxVersion:0.
> a=T38MaxBitRate:14400.
> a=T38FaxTranscodingMMR.
> a=T38FaxTranscodingJBIG.
> a=T38FaxRateManagement:transferredTCF.
> a=T38FaxMaxBuffer:500.
> a=T38FaxMaxDatagram:512.
> a=T38FaxUdpEC:t38UDPFEC.
>
>
> U 2012/08/23 15:34:24.093598 4.4.4.4:5060 -> 3.3.3.3:5060
> SIP/2.0 200 OK.
> From: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
> To: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
> Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
> CSeq: 32528365 INVITE.
> Server: DC-SIP/2.0.
> Contact: <sip:+888 at 4.4.4.4:5060;transport=udp>.
> Content-Type: application/sdp.
> Content-Length: 272.
> .
> v=0.
> o=- 3554717663 3554717664 IN IP4 4.4.4.4.
> s=-.
> c=IN IP4 4.4.4.4.
> t=0 0.
> m=image 22054 udptl t38.
> a=T38FaxVersion:0.
> a=T38MaxBitRate:14400.
> a=T38FaxRateManagement:transferredTCF.
> a=T38FaxMaxBuffer:72.
> a=T38FaxMaxDatagram:316.
> a=T38FaxUdpEC:t38UDPRedundancy.
>
>
> U 2012/08/23 15:34:33.885255 4.4.4.4:5060 -> 3.3.3.3:5060
> INVITE sip:gw+provider-voip-tel at 3.3.3.3:5060;transport=udp;gw=provider-voip-tel SIP/2.0.
> Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
> From: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
> To: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
> CSeq: 693734809 INVITE.
> Contact: <sip:+888 at 4.4.4.4:5060;transport=udp>.
> Content-Type: application/sdp.
> Content-Length: 272.
> .
> v=0.
> o=- 3554717663 3554717665 IN IP4 4.4.4.4.
> s=-.
> c=IN IP4 4.4.4.4.
> t=0 0.
> m=image 22054 udptl t38.
> a=T38FaxVersion:0.
> a=T38MaxBitRate:14400.
> a=T38FaxRateManagement:transferredTCF.
> a=T38FaxMaxBuffer:72.
> a=T38FaxMaxDatagram:316.
> a=T38FaxUdpEC:t38UDPRedundancy.
>
>
> U 2012/08/23 15:34:33.939092 3.3.3.3:5060 -> 4.4.4.4:5060
> SIP/2.0 200 OK.
> From: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
> To: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
> Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
> CSeq: 693734809 INVITE.
> Contact: <sip:gw+provider-voip-tel at 3.3.3.3:5060;transport=udp;gw=provider-voip-tel>.
> Accept: application/sdp.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 345.
> .
> v=0.
> o=FreeSWITCH 1345708964 1345708967 IN IP4 3.3.3.3.
> s=FreeSWITCH.
> c=IN IP4 3.3.3.3.
> t=0 0.
> m=image 19892 udptl t38.
> a=T38FaxVersion:0.
> a=T38MaxBitRate:14400.
> a=T38FaxTranscodingMMR.
> a=T38FaxTranscodingJBIG.
> a=T38FaxRateManagement:transferredTCF.
> a=T38FaxMaxBuffer:72.
> a=T38FaxMaxDatagram:512.
> a=T38FaxUdpEC:t38UDPRedundancy.
>
>
> U 2012/08/23 15:34:33.950201 3.3.3.3:5060 -> 4.4.4.4:5060
> INVITE sip:+888 at 4.4.4.4:5060;transport=udp SIP/2.0.
> From: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
> To: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
> Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
> CSeq: 32528366 INVITE.
> Contact: <sip:gw+provider-voip-tel at 3.3.3.3:5060;transport=udp;gw=provider-voip-tel>.
> User-Agent: c4-provider-ims-intx.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 338.
> X-Designated-Service: fax/t38.
> .
> v=0.
> o=FreeSWITCH 1345708964 1345708968 IN IP4 3.3.3.3.
> s=FreeSWITCH.
> c=IN IP4 3.3.3.3.
> t=0 0.
> m=image 19892 udptl t38.
> a=T38FaxVersion:0.
> a=T38MaxBitRate:14400.
> a=T38FaxTranscodingMMR.
> a=T38FaxTranscodingJBIG.
> a=T38FaxRateManagement:transferredTCF.
> a=T38FaxMaxBuffer:72.
> a=T38FaxMaxDatagram:512.
> a=T38FaxUdpEC:t38UDPFEC.
>
>
> U 2012/08/23 15:34:34.043165 4.4.4.4:5060 -> 3.3.3.3:5060
> BYE sip:gw+provider-voip-tel at 3.3.3.3:5060;transport=udp;gw=provider-voip-tel SIP/2.0.
> Call-ID: 3bf65bb6-ed27-11e1-a810-1d8ba1d695aa.
> From: <sip:+888 at 4.4.4.4>;tag=SDkvc5299-1.45.133.551104+1+a1b50004+4166b8ad.
> To: "+777" <sip:+777 at provider.voip.tel>;tag=70aprjj2Z1QKK.
> CSeq: 693734810 BYE.
> Content-Length: 0.
> Reason: Q.850; cause=41;text="Temporary failure".
> .
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Thu Aug 23 19:18:21 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 23 Aug 2012 10:18:21 -0500
Subject: [Freeswitch-users] RTP Timestamp changes after REFER - Question
 on RFC conformity
In-Reply-To: <CAAMQ8bRGeZm-7NWbRg4yoRZBqYTHEkXUCKUJJYgsEGZPZYNVgQ@mail.gmail.com>
References: <CAAMQ8bRGeZm-7NWbRg4yoRZBqYTHEkXUCKUJJYgsEGZPZYNVgQ@mail.gmail.com>
Message-ID: <CAKbxfG-6jiiH9HncmpAuPsknjmzgJntG_6eThBE6oYwckWtJ5Q@mail.gmail.com>

Do you have a pcap of it not setting the mark bit when the timestamp
changes because the code will always set the mark bit when the packet
is about to send is not exactly the next packet it expected to send
after the previous one/


On Thu, Aug 23, 2012 at 10:04 AM, Christian Benke <benkokakao at gmail.com> wrote:
> Hi!
>
> I had a problem with VOP Softclient(www.voiceoperatorpanel.com)  where
> the callee had one-way-audio after a call was REFERred to him. After
> lots of tracing, debugging and hairpulling i realized the problem lies
> in the changing rtp-timestamp when the RTP-stream is switched from the
> middleman to the initial call after the REFER. The Softclient was not
> able to cope with the changed timestamp and ignored the incoming
> RTP-packets, leading to no audio for the callee.
>
> I was eventually able to solve the problem by activating
> rtp-rewrite-timestamps on the profile(Also added it to
> http://wiki.freeswitch.org/wiki/RTP_Issues#Voiceoperatorpanel_VOP).
>
> However, i would like to know if FreeSWITCH/Sofia is working according
> to the RFC and if the Softclient is to blame for the problem(So i can
> file a bugreport with them).
>
> In this thread, Brian West states that  it's ok to skip forward in
> timestamps as long as the marker-bit is set:
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-July/060333.html
> However, the Marker-bit is not set by FreeSWITCH when the REFER occurs.
>
> I didn't find this stated in http://tools.ietf.org/html/rfc3550(But
> "timestamp" is mentioned a lot, so i may have missed it) but there's
> this bug-report for Asterisk, where the exact same problem is
> described and eventually handled:
> https://issues.asterisk.org/view.php?id=17007
>
> Could someone with more insight please elaborate?
>
> Best regards,
> Christian
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

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

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

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


From ben at fc-publishing.com  Thu Aug 23 18:56:57 2012
From: ben at fc-publishing.com (Ben Ringham)
Date: Thu, 23 Aug 2012 10:56:57 -0400
Subject: [Freeswitch-users] Unable to outbound with CounterPath eyebeam
Message-ID: <CADk0+Bchccn86Y=D5UMtT_KcaFWdTDAZZ+gw7U5EJnhrt7D1+Q@mail.gmail.com>

Hello all,
Maybe you can help with an issue I have had from about day 2 of my
fusion/freeswitch install.

I am able to receive calls inbound to my softphone (counterpath eyebeam
V1.5.19.4) When I try to dial out, or any extensions I am unable to
connect. Watching the CLI it the softphone seems to connect and talk with
freeswitch, then hangs up the call. Below is CLI of the call I am
attempting. Any input to correct this issue will greatly help.
Ben

+OK log level  [7]
freeswitch at internal> 2012-08-23 10:45:42.332691 [DEBUG] sofia.c:6466 IP
10.0.1.151 Approved by acl "domains[]". Access Granted.
2012-08-23 10:45:42.333693 [NOTICE] switch_channel.c:812 New Channel
sofia/internal/5555 at 10.0.1.162 [c915f249-a2d1-4c5e-8900-8ae47f3deea1]
2012-08-23 10:45:42.333693 [DEBUG] sofia.c:4744 Channel sofia/internal/
5555 at 10.0.1.162 entering state [received][100]
2012-08-23 10:45:42.333693 [DEBUG] sofia.c:4755 Remote SDP:
v=0
o=- 7 2 IN IP4 10.0.1.151
s=CounterPath eyeBeam 1.5
c=IN IP4 10.0.1.151
t=0 0
m=audio 51938 RTP/AVP 107 0 8 18 101
a=rtpmap:107 BV32/16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 3 : RQXbULH6 jeWr1bhp 10.0.1.151 51938
a=alt:2 2 : lIrXnsi1 yeNfshsQ 192.168.238.1 51938
a=alt:3 1 : np3FXIkK K342ICWN 192.168.64.1 51938

2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
[BV32:107:16000:20:0]/[G722:9:8000:20:64000]
2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
[BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
[BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
[BV32:107:16000:20:0]/[GSM:3:8000:20:13200]
2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-08-23 10:45:42.334691 [DEBUG] sofia_glue.c:2760 Set Codec
sofia/internal/5555 at 10.0.1.162 PCMU/8000 20 ms 160 samples 64000 bits
2012-08-23 10:45:42.334691 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/5555 at 10.0.1.162) Running State Change CS_NEW
2012-08-23 10:45:42.334691 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/5555 at 10.0.1.162) State NEW
2012-08-23 10:45:42.334691 [DEBUG] sofia_glue.c:4733 Set 2833 dtmf
send/recv payload to 101
2012-08-23 10:45:42.334691 [DEBUG] sofia.c:4922 (sofia/internal/
5555 at 10.0.1.162) State Change CS_NEW -> CS_INIT
2012-08-23 10:45:42.334691 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/5555 at 10.0.1.162 [BREAK]
2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/5555 at 10.0.1.162) Running State Change CS_INIT
2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:356
(sofia/internal/5555 at 10.0.1.162) State INIT
2012-08-23 10:45:42.335694 [DEBUG] mod_sofia.c:84 sofia/internal/
5555 at 10.0.1.162 SOFIA INIT
2012-08-23 10:45:42.335694 [DEBUG] mod_sofia.c:124 (sofia/internal/
5555 at 10.0.1.162) State Change CS_INIT -> CS_ROUTING
2012-08-23 10:45:42.335694 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/5555 at 10.0.1.162 [BREAK]
2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:356
(sofia/internal/5555 at 10.0.1.162) State INIT going to sleep
2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/5555 at 10.0.1.162) Running State Change CS_ROUTING
2012-08-23 10:45:42.335694 [DEBUG] switch_channel.c:1664 (sofia/internal/
5555 at 10.0.1.162) Callstate Change DOWN -> RINGING
2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:359
(sofia/internal/5555 at 10.0.1.162) State ROUTING
2012-08-23 10:45:42.335694 [DEBUG] mod_sofia.c:147 sofia/internal/
5555 at 10.0.1.162 SOFIA ROUTING
2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:77
sofia/internal/5555 at 10.0.1.162 Standard ROUTING
2012-08-23 10:45:42.335694 [INFO] mod_dialplan_xml.c:331 Processing Ben
<5555>->18002267611 in context public
Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->unloop]
continue=false
Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->outside_call]
continue=true
Dialplan: sofia/internal/5555 at 10.0.1.162 Absolute Condition [outside_call]
Dialplan: sofia/internal/5555 at 10.0.1.162 Action set(outside_call=true)
Dialplan: sofia/internal/5555 at 10.0.1.162 Action
set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->call_debug]
continue=true
Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL) [call_debug]
${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/5555 at 10.0.1.162 parsing
[public->public_extensions] continue=false
Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL) [public_extensions]
destination_number(18002267611) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->inbound to ext
55] continue=false
Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (PASS) [inbound to ext 55]
context(public) =~ /public/ break=on-false
Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL) [inbound to ext 55]
destination_number(18002267611) =~ /4049371797/ break=on-false
2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:119
(sofia/internal/5555 at 10.0.1.162) State Change CS_ROUTING -> CS_EXECUTE
2012-08-23 10:45:42.337693 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/5555 at 10.0.1.162 [BREAK]
2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:359
(sofia/internal/5555 at 10.0.1.162) State ROUTING going to sleep
2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/5555 at 10.0.1.162) Running State Change CS_EXECUTE
2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:366
(sofia/internal/5555 at 10.0.1.162) State EXECUTE
2012-08-23 10:45:42.337693 [DEBUG] mod_sofia.c:240 sofia/internal/
5555 at 10.0.1.162 SOFIA EXECUTE
2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:157
sofia/internal/5555 at 10.0.1.162 Standard EXECUTE
EXECUTE sofia/internal/5555 at 10.0.1.162 set(outside_call=true)
2012-08-23 10:45:42.337693 [DEBUG] mod_dptools.c:1059 sofia/internal/
5555 at 10.0.1.162 SET [outside_call]=[true]
EXECUTE sofia/internal/5555 at 10.0.1.162 set(RFC2822_DATE=Thu, 23 Aug 2012
10:45:42 +0530)
2012-08-23 10:45:42.338693 [DEBUG] mod_dptools.c:1059 sofia/internal/
5555 at 10.0.1.162 SET [RFC2822_DATE]=[Thu, 23 Aug 2012 10:45:42 +0530]
2012-08-23 10:45:42.338693 [NOTICE] switch_core_state_machine.c:189
sofia/internal/5555 at 10.0.1.162 has executed the last dialplan instruction,
hanging up.
2012-08-23 10:45:42.338693 [DEBUG] switch_channel.c:2559 (sofia/internal/
5555 at 10.0.1.162) Callstate Change RINGING -> HANGUP
2012-08-23 10:45:42.338693 [NOTICE] switch_core_state_machine.c:191 Hangup
sofia/internal/5555 at 10.0.1.162 [CS_EXECUTE] [NORMAL_CLEARING]
2012-08-23 10:45:42.338693 [DEBUG] switch_channel.c:2575 Send signal
sofia/internal/5555 at 10.0.1.162 [KILL]
2012-08-23 10:45:42.338693 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/5555 at 10.0.1.162 [BREAK]
2012-08-23 10:45:42.338693 [DEBUG] switch_core_state_machine.c:366
(sofia/internal/5555 at 10.0.1.162) State EXECUTE going to sleep
2012-08-23 10:45:42.338693 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/5555 at 10.0.1.162) Running State Change CS_HANGUP
2012-08-23 10:45:42.339694 [DEBUG] switch_core_state_machine.c:560
(sofia/internal/5555 at 10.0.1.162) State HANGUP
2012-08-23 10:45:42.339694 [DEBUG] mod_sofia.c:457 Channel sofia/internal/
5555 at 10.0.1.162 hanging up, cause: NORMAL_CLEARING
2012-08-23 10:45:42.339694 [DEBUG] mod_sofia.c:519 Responding to INVITE
with: 480
2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:46
sofia/internal/5555 at 10.0.1.162 Standard HANGUP, cause: NORMAL_CLEARING
2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:560
(sofia/internal/5555 at 10.0.1.162) State HANGUP going to sleep
2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/5555 at 10.0.1.162) State Change CS_HANGUP -> CS_REPORTING
2012-08-23 10:45:42.340695 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/5555 at 10.0.1.162 [BREAK]
2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/5555 at 10.0.1.162) Running State Change CS_REPORTING
2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:620
(sofia/internal/5555 at 10.0.1.162) State REPORTING
2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:53
sofia/internal/5555 at 10.0.1.162 Standard REPORTING, cause: NORMAL_CLEARING
2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:620
(sofia/internal/5555 at 10.0.1.162) State REPORTING going to sleep
2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:345
(sofia/internal/5555 at 10.0.1.162) State Change CS_REPORTING -> CS_DESTROY
2012-08-23 10:45:42.489418 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/5555 at 10.0.1.162 [BREAK]
2012-08-23 10:45:42.489418 [DEBUG] switch_core_session.c:1288 Session 22
(sofia/internal/5555 at 10.0.1.162) Locked, Waiting on external entities
2012-08-23 10:45:42.489418 [NOTICE] switch_core_session.c:1306 Session 22
(sofia/internal/5555 at 10.0.1.162) Ended
2012-08-23 10:45:42.489418 [NOTICE] switch_core_session.c:1308 Close
Channel sofia/internal/5555 at 10.0.1.162 [CS_DESTROY]
2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:449
(sofia/internal/5555 at 10.0.1.162) Callstate Change HANGUP -> DOWN
2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:452
(sofia/internal/5555 at 10.0.1.162) Running State Change CS_DESTROY
2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:462
(sofia/internal/5555 at 10.0.1.162) State DESTROY
2012-08-23 10:45:42.491481 [DEBUG] mod_sofia.c:362 sofia/internal/
5555 at 10.0.1.162 SOFIA DESTROY
2012-08-23 10:45:42.491481 [DEBUG] switch_core_state_machine.c:60
sofia/internal/5555 at 10.0.1.162 Standard DESTROY
2012-08-23 10:45:42.491481 [DEBUG] switch_core_state_machine.c:462
(sofia/internal/5555 at 10.0.1.162) State DESTROY going to sleep
-------------- next part --------------
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From benkokakao at gmail.com  Thu Aug 23 19:31:31 2012
From: benkokakao at gmail.com (Christian Benke)
Date: Thu, 23 Aug 2012 17:31:31 +0200
Subject: [Freeswitch-users] RTP Timestamp changes after REFER - Question
 on RFC conformity
In-Reply-To: <CAKbxfG-6jiiH9HncmpAuPsknjmzgJntG_6eThBE6oYwckWtJ5Q@mail.gmail.com>
References: <CAAMQ8bRGeZm-7NWbRg4yoRZBqYTHEkXUCKUJJYgsEGZPZYNVgQ@mail.gmail.com>
	<CAKbxfG-6jiiH9HncmpAuPsknjmzgJntG_6eThBE6oYwckWtJ5Q@mail.gmail.com>
Message-ID: <CAAMQ8bReGJZdZeeiNeq5_38_Y=wrCd-8ARLzAgZe8kymyJ-gzQ@mail.gmail.com>

Hi!

Here's a trace: http://poab.org/misc/one_way_audio_internal_g711.pcap
(or http://filedump.org/files/yMnIRQUC1345735448.html for the archive)

The REFER happens at Paket 1305, Time 25.662729

The RTP-Paket that are sent to the C-endpoint after the REFER are
continuous from A's point of view(As it is the continuation of the
A-B-stream), but not from C's point of view.

Regards,
Christian

On 23 August 2012 17:18, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> Do you have a pcap of it not setting the mark bit when the timestamp
> changes because the code will always set the mark bit when the packet
> is about to send is not exactly the next packet it expected to send
> after the previous one/
>
>
> On Thu, Aug 23, 2012 at 10:04 AM, Christian Benke <benkokakao at gmail.com> wrote:
>> Hi!
>>
>> I had a problem with VOP Softclient(www.voiceoperatorpanel.com)  where
>> the callee had one-way-audio after a call was REFERred to him. After
>> lots of tracing, debugging and hairpulling i realized the problem lies
>> in the changing rtp-timestamp when the RTP-stream is switched from the
>> middleman to the initial call after the REFER. The Softclient was not
>> able to cope with the changed timestamp and ignored the incoming
>> RTP-packets, leading to no audio for the callee.
>>
>> I was eventually able to solve the problem by activating
>> rtp-rewrite-timestamps on the profile(Also added it to
>> http://wiki.freeswitch.org/wiki/RTP_Issues#Voiceoperatorpanel_VOP).
>>
>> However, i would like to know if FreeSWITCH/Sofia is working according
>> to the RFC and if the Softclient is to blame for the problem(So i can
>> file a bugreport with them).
>>
>> In this thread, Brian West states that  it's ok to skip forward in
>> timestamps as long as the marker-bit is set:
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-July/060333.html
>> However, the Marker-bit is not set by FreeSWITCH when the REFER occurs.
>>
>> I didn't find this stated in http://tools.ietf.org/html/rfc3550(But
>> "timestamp" is mentioned a lot, so i may have missed it) but there's
>> this bug-report for Asterisk, where the exact same problem is
>> described and eventually handled:
>> https://issues.asterisk.org/view.php?id=17007
>>
>> Could someone with more insight please elaborate?
>>
>> Best regards,
>> Christian
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Thu Aug 23 19:38:34 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Aug 2012 08:38:34 -0700
Subject: [Freeswitch-users] Dial out within a conference with early media
In-Reply-To: <EC5B3D9E-68E3-4974-AA17-EFE65D96CC52@kavun.ch>
References: <EC5B3D9E-68E3-4974-AA17-EFE65D96CC52@kavun.ch>
Message-ID: <CAKzWOxUnmh8LW=YAHvkJ8GoHP+mG0xs2vkhuiuguL0rt9iBCug@mail.gmail.com>

For item #5 - is that a SIP INFO message or is that some audio saying that
the user is not available? If it's early media it should cause the dial to
be "successful" unless you're explicitly setting ignore_early_media=true in
your dialstring.

-MC

On Wed, Aug 22, 2012 at 5:20 PM, Emrah <lists at kavun.ch> wrote:

> Hi guys,
>
> Is there an option to dial with early media support from within a
> conference?
>
> This is what happens now:
>
> 1. I have an active conf.
> 2. I decide to pull in an additional user by calling her mobile.
> 3. From fs_cli, I execute conference <conf-id> dial
> Sofia/gateway/xyz/1234567890. Then
> 4. Either she picks up, we her the enter sound and we know she's joined in.
> 5. Or her mobile is switched off, with no voicemail and just an INFO
> message... And we will never know what happened.
>
> Is there a way to accomplish the following?
>
> Have the conference broadcast the early media portion of the call, and
> still play the enter sound when the called party picks up?
>
> If this is not possible, I think it would be a great addition to the
> conferencing module of FS to have a dedicated dial command with support for
> Early Media.
>
> Best and thanks,
> Emrah
> --
> Emrah
>
> ?In theory, theory and practice are the same. In practice, they are
> not.? Albert Einstein
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Aug 23 19:40:15 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Aug 2012 08:40:15 -0700
Subject: [Freeswitch-users] Polycom firmware and TLS
In-Reply-To: <CAKV-jFwYbr9O=EH=_kQsUOqOnY0xHRqyBn_SmHTcQkC-pkyxZw@mail.gmail.com>
References: <CAKDfjgeZe7opEJGwZTuJNhTt-uBHLaLfD-Ckx7HYRskk9chk4Q@mail.gmail.com>
	<CAKzWOxU5c2YbhELsyKnUt=RFVod5bLqeRUUkVMdxoJEC_freHA@mail.gmail.com>
	<CAKV-jFwYbr9O=EH=_kQsUOqOnY0xHRqyBn_SmHTcQkC-pkyxZw@mail.gmail.com>
Message-ID: <CAKzWOxUAATzykXkGGL1j4HAHQE8aJTK2Kvxmfz8Vvf8itD1NsQ@mail.gmail.com>

You know, this might be a good topic for the Wednesday conference calls.
I'm a bit green with the whole thing with certificates, CA, self-signed
certs, etc. Is there someone who is comfortable giving us a tutorial on all
this?

-MC

On Wed, Aug 22, 2012 at 4:38 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> I have a phone running 4.0.2 ish Polycom UC firmware. I might have to go
> ahead and bite the bullet and test. I don't have much experience in the way
> of TLS, so it's time to pull my finger out and start learning.
>
> - BDF
>
> On Wed, Aug 22, 2012 at 12:04 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>>  On Wed, Aug 22, 2012 at 8:58 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:
>>
>>> All,
>>>
>>>   Which versions of Polycom firmware have you had the most success
>>> using TLS with?
>>>
>>> Thanks!
>>>
>>
>> I'm interested in this as well. I believe our interop pages haven't had
>> much in the way of updates lately and all the major MFGs have released new
>> versions of their firmware. It would be nice to know what works and what
>> doesn't.
>> -MC
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From bdfoster at endigotech.com  Thu Aug 23 19:50:51 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 23 Aug 2012 11:50:51 -0400
Subject: [Freeswitch-users] Unable to outbound with CounterPath eyebeam
In-Reply-To: <CADk0+Bchccn86Y=D5UMtT_KcaFWdTDAZZ+gw7U5EJnhrt7D1+Q@mail.gmail.com>
References: <CADk0+Bchccn86Y=D5UMtT_KcaFWdTDAZZ+gw7U5EJnhrt7D1+Q@mail.gmail.com>
Message-ID: <CAKV-jFx9_-BVn=3JoP9QVSWhQ6VtbN576y6==Okxg5gv_8Q0cQ@mail.gmail.com>

We've been through this before. Why are you on the public context with your
internal extension? You should be on the default context. It's not doing
anything because nothing is matching, and when that happens the call is
hung up (with NORMAL CLEARNING).

-BDF

On Thu, Aug 23, 2012 at 10:56 AM, Ben Ringham <ben at fc-publishing.com> wrote:

> Hello all,
> Maybe you can help with an issue I have had from about day 2 of my
> fusion/freeswitch install.
>
> I am able to receive calls inbound to my softphone (counterpath eyebeam
> V1.5.19.4) When I try to dial out, or any extensions I am unable to
> connect. Watching the CLI it the softphone seems to connect and talk with
> freeswitch, then hangs up the call. Below is CLI of the call I am
> attempting. Any input to correct this issue will greatly help.
> Ben
>
> +OK log level  [7]
> freeswitch at internal> 2012-08-23 10:45:42.332691 [DEBUG] sofia.c:6466 IP
> 10.0.1.151 Approved by acl "domains[]". Access Granted.
> 2012-08-23 10:45:42.333693 [NOTICE] switch_channel.c:812 New Channel
> sofia/internal/5555 at 10.0.1.162 [c915f249-a2d1-4c5e-8900-8ae47f3deea1]
> 2012-08-23 10:45:42.333693 [DEBUG] sofia.c:4744 Channel sofia/internal/
> 5555 at 10.0.1.162 entering state [received][100]
> 2012-08-23 10:45:42.333693 [DEBUG] sofia.c:4755 Remote SDP:
> v=0
> o=- 7 2 IN IP4 10.0.1.151
> s=CounterPath eyeBeam 1.5
> c=IN IP4 10.0.1.151
> t=0 0
> m=audio 51938 RTP/AVP 107 0 8 18 101
> a=rtpmap:107 BV32/16000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=yes
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=alt:1 3 : RQXbULH6 jeWr1bhp 10.0.1.151 51938
> a=alt:2 2 : lIrXnsi1 yeNfshsQ 192.168.238.1 51938
> a=alt:3 1 : np3FXIkK K342ICWN 192.168.64.1 51938
>
> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
> [BV32:107:16000:20:0]/[G722:9:8000:20:64000]
> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
> [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
> [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
> [BV32:107:16000:20:0]/[GSM:3:8000:20:13200]
> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2012-08-23 10:45:42.334691 [DEBUG] sofia_glue.c:2760 Set Codec
> sofia/internal/5555 at 10.0.1.162 PCMU/8000 20 ms 160 samples 64000 bits
> 2012-08-23 10:45:42.334691 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_NEW
> 2012-08-23 10:45:42.334691 [DEBUG] switch_core_state_machine.c:338
> (sofia/internal/5555 at 10.0.1.162) State NEW
> 2012-08-23 10:45:42.334691 [DEBUG] sofia_glue.c:4733 Set 2833 dtmf
> send/recv payload to 101
> 2012-08-23 10:45:42.334691 [DEBUG] sofia.c:4922 (sofia/internal/
> 5555 at 10.0.1.162) State Change CS_NEW -> CS_INIT
> 2012-08-23 10:45:42.334691 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/5555 at 10.0.1.162 [BREAK]
> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_INIT
> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:356
> (sofia/internal/5555 at 10.0.1.162) State INIT
> 2012-08-23 10:45:42.335694 [DEBUG] mod_sofia.c:84 sofia/internal/
> 5555 at 10.0.1.162 SOFIA INIT
> 2012-08-23 10:45:42.335694 [DEBUG] mod_sofia.c:124 (sofia/internal/
> 5555 at 10.0.1.162) State Change CS_INIT -> CS_ROUTING
> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/5555 at 10.0.1.162 [BREAK]
> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:356
> (sofia/internal/5555 at 10.0.1.162) State INIT going to sleep
> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_ROUTING
> 2012-08-23 10:45:42.335694 [DEBUG] switch_channel.c:1664 (sofia/internal/
> 5555 at 10.0.1.162) Callstate Change DOWN -> RINGING
> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:359
> (sofia/internal/5555 at 10.0.1.162) State ROUTING
> 2012-08-23 10:45:42.335694 [DEBUG] mod_sofia.c:147 sofia/internal/
> 5555 at 10.0.1.162 SOFIA ROUTING
> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:77
> sofia/internal/5555 at 10.0.1.162 Standard ROUTING
> 2012-08-23 10:45:42.335694 [INFO] mod_dialplan_xml.c:331 Processing Ben
> <5555>->18002267611 in context public
> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->unloop]
> continue=false
> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->outside_call]
> continue=true
> Dialplan: sofia/internal/5555 at 10.0.1.162 Absolute Condition [outside_call]
> Dialplan: sofia/internal/5555 at 10.0.1.162 Action set(outside_call=true)
> Dialplan: sofia/internal/5555 at 10.0.1.162 Action
> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->call_debug]
> continue=true
> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL) [call_debug]
> ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing
> [public->public_extensions] continue=false
> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL) [public_extensions]
> destination_number(18002267611) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->inbound to ext
> 55] continue=false
> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (PASS) [inbound to ext 55]
> context(public) =~ /public/ break=on-false
> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL) [inbound to ext 55]
> destination_number(18002267611) =~ /4049371797/ break=on-false
> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:119
> (sofia/internal/5555 at 10.0.1.162) State Change CS_ROUTING -> CS_EXECUTE
> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/5555 at 10.0.1.162 [BREAK]
> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:359
> (sofia/internal/5555 at 10.0.1.162) State ROUTING going to sleep
> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_EXECUTE
> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:366
> (sofia/internal/5555 at 10.0.1.162) State EXECUTE
> 2012-08-23 10:45:42.337693 [DEBUG] mod_sofia.c:240 sofia/internal/
> 5555 at 10.0.1.162 SOFIA EXECUTE
> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:157
> sofia/internal/5555 at 10.0.1.162 Standard EXECUTE
> EXECUTE sofia/internal/5555 at 10.0.1.162 set(outside_call=true)
> 2012-08-23 10:45:42.337693 [DEBUG] mod_dptools.c:1059 sofia/internal/
> 5555 at 10.0.1.162 SET [outside_call]=[true]
> EXECUTE sofia/internal/5555 at 10.0.1.162 set(RFC2822_DATE=Thu, 23 Aug 2012
> 10:45:42 +0530)
> 2012-08-23 10:45:42.338693 [DEBUG] mod_dptools.c:1059 sofia/internal/
> 5555 at 10.0.1.162 SET [RFC2822_DATE]=[Thu, 23 Aug 2012 10:45:42 +0530]
> 2012-08-23 10:45:42.338693 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/5555 at 10.0.1.162 has executed the last dialplan
> instruction, hanging up.
> 2012-08-23 10:45:42.338693 [DEBUG] switch_channel.c:2559 (sofia/internal/
> 5555 at 10.0.1.162) Callstate Change RINGING -> HANGUP
> 2012-08-23 10:45:42.338693 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/internal/5555 at 10.0.1.162 [CS_EXECUTE] [NORMAL_CLEARING]
> 2012-08-23 10:45:42.338693 [DEBUG] switch_channel.c:2575 Send signal
> sofia/internal/5555 at 10.0.1.162 [KILL]
> 2012-08-23 10:45:42.338693 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/5555 at 10.0.1.162 [BREAK]
> 2012-08-23 10:45:42.338693 [DEBUG] switch_core_state_machine.c:366
> (sofia/internal/5555 at 10.0.1.162) State EXECUTE going to sleep
> 2012-08-23 10:45:42.338693 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_HANGUP
> 2012-08-23 10:45:42.339694 [DEBUG] switch_core_state_machine.c:560
> (sofia/internal/5555 at 10.0.1.162) State HANGUP
> 2012-08-23 10:45:42.339694 [DEBUG] mod_sofia.c:457 Channel sofia/internal/
> 5555 at 10.0.1.162 hanging up, cause: NORMAL_CLEARING
> 2012-08-23 10:45:42.339694 [DEBUG] mod_sofia.c:519 Responding to INVITE
> with: 480
> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/5555 at 10.0.1.162 Standard HANGUP, cause: NORMAL_CLEARING
> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:560
> (sofia/internal/5555 at 10.0.1.162) State HANGUP going to sleep
> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:351
> (sofia/internal/5555 at 10.0.1.162) State Change CS_HANGUP -> CS_REPORTING
> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/5555 at 10.0.1.162 [BREAK]
> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:320
> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_REPORTING
> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:620
> (sofia/internal/5555 at 10.0.1.162) State REPORTING
> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/5555 at 10.0.1.162 Standard REPORTING, cause: NORMAL_CLEARING
> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:620
> (sofia/internal/5555 at 10.0.1.162) State REPORTING going to sleep
> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:345
> (sofia/internal/5555 at 10.0.1.162) State Change CS_REPORTING -> CS_DESTROY
> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/5555 at 10.0.1.162 [BREAK]
> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_session.c:1288 Session 22
> (sofia/internal/5555 at 10.0.1.162) Locked, Waiting on external entities
> 2012-08-23 10:45:42.489418 [NOTICE] switch_core_session.c:1306 Session 22
> (sofia/internal/5555 at 10.0.1.162) Ended
> 2012-08-23 10:45:42.489418 [NOTICE] switch_core_session.c:1308 Close
> Channel sofia/internal/5555 at 10.0.1.162 [CS_DESTROY]
> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:449
> (sofia/internal/5555 at 10.0.1.162) Callstate Change HANGUP -> DOWN
> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:452
> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_DESTROY
> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:462
> (sofia/internal/5555 at 10.0.1.162) State DESTROY
> 2012-08-23 10:45:42.491481 [DEBUG] mod_sofia.c:362 sofia/internal/
> 5555 at 10.0.1.162 SOFIA DESTROY
> 2012-08-23 10:45:42.491481 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/5555 at 10.0.1.162 Standard DESTROY
> 2012-08-23 10:45:42.491481 [DEBUG] switch_core_state_machine.c:462
> (sofia/internal/5555 at 10.0.1.162) State DESTROY going to sleep
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From msc at freeswitch.org  Thu Aug 23 19:51:22 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Aug 2012 08:51:22 -0700
Subject: [Freeswitch-users] Voice broadcasting
In-Reply-To: <CAPoPLj_QVqmA43PWM9rpzWrAii3k_3kp4BhYxngip+A84NbQAw@mail.gmail.com>
References: <CAAdTMa-Qz1UyH+wXKEGXRBHF=Hjh7S_jpxe_BSphXgsrYpo_aw@mail.gmail.com>
	<CC359325.3A8AE%krice@freeswitch.org>
	<CAKV-jFzQizHQAbUEQxCPdi_Wc9odhcL0j7oc9grHX=kceZ3-aQ@mail.gmail.com>
	<CAAdTMa-wwTiyZ+=caecU8mC5K8Z2A_2Atnq4yonDRz-172BU5g@mail.gmail.com>
	<CAOL0t+LYhOBid83cWPwvUEPqyE309Vm6DC7ZkoQWO42CVUp4Og@mail.gmail.com>
	<CAKV-jFykvFmKSPf1ZGL0tdth2ZJPyk2tX0J2aiWJHYW5Zg7vRA@mail.gmail.com>
	<CAPoPLj_QVqmA43PWM9rpzWrAii3k_3kp4BhYxngip+A84NbQAw@mail.gmail.com>
Message-ID: <CAKzWOxU8qtor5ZkENWdncGp89YP7vFHSQJCxD03y3zK-+chM0w@mail.gmail.com>

On Thu, Aug 23, 2012 at 5:37 AM, David J <david at styleflare.com> wrote:

> Brian.
>
> I will agree with you only if ICT dialer is a commercial product.
>
> Otherwise to mention it on list should be OK as long as its free switch
> related.
>
We are pretty tolerant of people mentioning their FreeSWITCH-based
commercial products. However, we don't want the list to turn into a
marketing free-for-all. If you have a commercial product that you think
would meet the OP's needs then it's okay to mention it. The caveat is that
you really must also list any [FL]OSS options as well, even if those free
options are less complete than your own commercial solution.

Here's an example:

Q: "What's the best GUI for FreeSWITCH?"

A: "IMHO, the best GUI is the CudaTel. More info available at
www.cudatel.com. If you are looking for an OSS alternative you can check
out Blue.Box (2600hz.org) or FusionPBX (fusionpbx.com) and try those out.
I've also heard of FreePyBX (freepybx.org) but I know very little about
that one."

If you keep it balanced, reasonable, and non-spammy then you'll be fine. :)

-MC
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From avi at avimarcus.net  Thu Aug 23 19:57:34 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 23 Aug 2012 18:57:34 +0300
Subject: [Freeswitch-users] Voice broadcasting
In-Reply-To: <CAKzWOxU8qtor5ZkENWdncGp89YP7vFHSQJCxD03y3zK-+chM0w@mail.gmail.com>
References: <CAAdTMa-Qz1UyH+wXKEGXRBHF=Hjh7S_jpxe_BSphXgsrYpo_aw@mail.gmail.com>
	<CC359325.3A8AE%krice@freeswitch.org>
	<CAKV-jFzQizHQAbUEQxCPdi_Wc9odhcL0j7oc9grHX=kceZ3-aQ@mail.gmail.com>
	<CAAdTMa-wwTiyZ+=caecU8mC5K8Z2A_2Atnq4yonDRz-172BU5g@mail.gmail.com>
	<CAOL0t+LYhOBid83cWPwvUEPqyE309Vm6DC7ZkoQWO42CVUp4Og@mail.gmail.com>
	<CAKV-jFykvFmKSPf1ZGL0tdth2ZJPyk2tX0J2aiWJHYW5Zg7vRA@mail.gmail.com>
	<CAPoPLj_QVqmA43PWM9rpzWrAii3k_3kp4BhYxngip+A84NbQAw@mail.gmail.com>
	<CAKzWOxU8qtor5ZkENWdncGp89YP7vFHSQJCxD03y3zK-+chM0w@mail.gmail.com>
Message-ID: <CALNZuwkobxi_6pVf14HH6P7MfMTD7eFLuP4tYmrPQvH6R7JiDQ@mail.gmail.com>

Since ICT Dialer was mentioned only on two threads where it was related,
and it is indeed OSS, it seems entirely appropriate.

Their download page http://www.ictdialer.org/node/7 ->
https://github.com/ictinnovations/ictdialer

-Avi


On Thu, Aug 23, 2012 at 6:51 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Aug 23, 2012 at 5:37 AM, David J <david at styleflare.com> wrote:
>
>> Brian.
>>
>> I will agree with you only if ICT dialer is a commercial product.
>>
>> Otherwise to mention it on list should be OK as long as its free switch
>> related.
>>
> We are pretty tolerant of people mentioning their FreeSWITCH-based
> commercial products. However, we don't want the list to turn into a
> marketing free-for-all. If you have a commercial product that you think
> would meet the OP's needs then it's okay to mention it. The caveat is that
> you really must also list any [FL]OSS options as well, even if those free
> options are less complete than your own commercial solution.
>
> Here's an example:
>
> Q: "What's the best GUI for FreeSWITCH?"
>
> A: "IMHO, the best GUI is the CudaTel. More info available at
> www.cudatel.com. If you are looking for an OSS alternative you can check
> out Blue.Box (2600hz.org) or FusionPBX (fusionpbx.com) and try those out.
> I've also heard of FreePyBX (freepybx.org) but I know very little about
> that one."
>
> If you keep it balanced, reasonable, and non-spammy then you'll be fine. :)
>
> -MC
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 asaad2 at gmail.com  Thu Aug 23 20:01:15 2012
From: asaad2 at gmail.com (BookBag)
Date: Thu, 23 Aug 2012 12:01:15 -0400
Subject: [Freeswitch-users] FreeSWITCH SIP trunk with CUCM 7.1
In-Reply-To: <CAKV-jFzU+908ZpxC0ap29UJ=-rfrQ93fUB_aTMfYHNYL8LGO6Q@mail.gmail.com>
References: <CAHwAnSFyCQv_dELUKFe_jBmAYcvetw1uLOFbZ-cioZzCuZhi=w@mail.gmail.com>
	<CAAdTMa-Tjrxq6UUcosFN=f7pNPXqTZXnu3tL_uf6ChAwQ5ezRQ@mail.gmail.com>
	<CAHwAnSEWoZyxBosP=MVjVctzi3_wAioUsv3_9kZEhD_yoWomJA@mail.gmail.com>
	<CAKV-jFzU+908ZpxC0ap29UJ=-rfrQ93fUB_aTMfYHNYL8LGO6Q@mail.gmail.com>
Message-ID: <CAAdTMa-CHvFtpp35WN5sZVvVJXuQDBfBN_RJRBKDzticqvDurQ@mail.gmail.com>

Install one cucm as publisher and after that finishes install the second
one as subscriber. If you want FS to be your third failover my guess is
that your going to need to edit your router config to forward your phones
to FS since your cucm is the only thing directing your calls to FS
On Aug 23, 2012 7:10 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> Use OpenSIPS.
>
> Brian Foster
> Endigo Computer LLC
>
> Sent from a mobile device.
> On Aug 23, 2012 3:39 AM, "Anton Vojlenko" <stargray at bigmir.net> wrote:
>
>> We have two CUCMs and I need create failover with them. How can I do
>> this?
>>
>> 2012/8/23 BookBag <asaad2 at gmail.com>
>>
>>> I actually have that running. I configured a sip trunk between fs and
>>> cucm 7.1.3.  The trick is avoid sip authentication at all costs. Cucm does
>>> not do sip authentication. It can neither accept or register to another sip
>>> trunk.
>>>
>>> Two things need to be done. First on cucm create a sip trunk to fs using
>>> ip address and in your dialplan establish a sip dial route with a prefix
>>> like for example 81. So all cisco users who dial 81xxxx will be routed
>>> through that trunk to your fs ipaddress.
>>>
>>> The other thing that you do is edit your acl to accept everything from
>>> your cucm ip address. Then in your dial plan establish another prefix for
>>> example for example 71. So all fs users will dial 71xxxx to get to a cucm
>>> extension will go directly to xxxx@<cucm ip>.  Let me know if you need
>>> additional help.
>>> On Aug 22, 2012 12:03 PM, "????? ????????" <stargray at bigmir.net> wrote:
>>>
>>>> Good day!
>>>>
>>>> We have next task. We need config SIP trunk between FreeSWITCH and CUCM
>>>> 7.1. FreeSWITCH have to register on CUCM.
>>>>
>>>> We have made in CUCM SIP trunk with correct SIP Trunk Security Profilefor authorization and created Application User.
>>>>
>>>> But FreeSWITCH can not registered. There is 405 error "SIP trunk
>>>> disallow REGISTER". See attachments.
>>>>
>>>> Maybe someone faced with a similar problem?
>>>>
>>>>
>>>> Thanks,
>>>> Anton
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Thu Aug 23 20:04:11 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 23 Aug 2012 09:04:11 -0700
Subject: [Freeswitch-users] FreeSWITCH and ISDN
In-Reply-To: <CADnUQS_eejGVz3a47EX2jO9d5C7zf-o03HTuimi3Sbm=kPJrSw@mail.gmail.com>
References: <CADnUQS_eejGVz3a47EX2jO9d5C7zf-o03HTuimi3Sbm=kPJrSw@mail.gmail.com>
Message-ID: <CAKzWOxWdra3sXuCZ_0KSn2GmUfmD85pNxiLWmT3xFGZYSC=5Fw@mail.gmail.com>

Nowadays the whole Digium vs. Sangoma thing has boiled down to preference
more than anything else. They are both pretty reliable (I've used both) so
I wouldn't use that metric as a tie-breaker. If money is the ultimate
deciding factor then a Digium clone is probably the cheapest. However, if
you want to support the company who has supported the FreeSWITCH community
the most then it's hands down Sangoma. Also, we know all the Sangoma guys
personally (Hi Moy!) and they have produced the bulk of mod_freetdm. Buying
Sangoma is a great way to support FreeSWITCH.

Just my opinion, but I hope it's good food for thought.
-MC

On Thu, Aug 23, 2012 at 1:42 AM, Tor Petterson <tpe at actimizer.com> wrote:

> Hi
>
> I have a FreeSWITCH installation, doing outbound telephony, with a
> peak load of about 500 to 600 concurrent channels.
> Now I have to do a similar setup in a market where SIP telephony is
> not available, so it has to be ISDN.
> I am looking at buying either Digium or Sangoma cards. Does anyone
> have any experiences about reliability of these cards?
> How does the cpu load per channel compare to SIP telephony?
>
> --
> Tor Petterson
>
> tpe at actimizer.com
>
> Tobaksvejen 25, 2. tv. - 2860 S?borg
> Telephone: +45 39 55 05 32
> www.actimizer.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From mitch.capper at gmail.com  Thu Aug 23 20:14:58 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 23 Aug 2012 09:14:58 -0700
Subject: [Freeswitch-users] TLS documentation inaccuracy
In-Reply-To: <5036220C.9080203@gmail.com>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in> <503226CE.7040307@hw.ac.uk>
	<50323D6A.3020708@anew.com.ve> <2526639.vT3cT5qGKG@virtex>
	<5033C5B1.3070509@hw.ac.uk> <5036220C.9080203@gmail.com>
Message-ID: <CALn233pgL5V1itSiydhX0vV8xVZXzTKgVFD+3+W6dCRAU3x+xQ@mail.gmail.com>

Vadim update your copy of FS.  The copy you are using is very old so
the gentls usage has changed 1.2 or git head should give you no
problem.   A .crt file contains only the certificate the .pem has the
certificate and private key, just copy the certificate part (its a
text document) to a file by itself and call .crt to get the .crt.

~Mitch

On Thu, Aug 23, 2012 at 5:29 AM, vaad.fabi at gmail.com
<vaad.fabi at gmail.com> wrote:
>   Hi all,
>
> There are some TLS documentation inaccuracy
> (http://wiki.freeswitch.org/wiki/Tls):
>
> 1.  " ./gentls_cert create_server -cn pbx.freeswitch.org -alt
> DNS:pbx.freeswitch.org -org freeswitch.org"
> There is no option "create_server" for gentls_cert , as i can see only
> setup,create,remove.
>
> 2.  "./gentls_cert create_client -cn Client1 -out Client1"
> There is no option "create_client" for gentls_cert.
> And there is no exactly information for softphone certificate
> generation. For Windows we need .crt file certificate, but how it could
> be generated (.pem only generated) ???
>
> Please fix doc and gentls_cert script, or explain all procedures. Thx.
>
> --
> Best Regards,
> Vadim F.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 fdelawarde at wirelessmundi.com  Thu Aug 23 20:28:24 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Thu, 23 Aug 2012 18:28:24 +0200
Subject: [Freeswitch-users] sip-router or opensips for good interoperability
	with freeswitch?
Message-ID: <1345739304.23436.89.camel@luna.madrid.commsmundi.com>

Hi,

Which of opensips or sip-router (kamailio) do you guys recommend to use
in front of freeswitch to create some big reliable cluster?

I'm not sure if there is any huge difference as both seem to share a
same origin. Should I just throw a dice?, or is there one that is
clearly better than the other (or one I should avoid) in terms of
reliability / performance, or friendliness with FS?

Thanks,
-- 
Fran?ois Delawarde <fdelawarde at wirelessmundi.com>
Wireless Mundi S.L.



From ben at fc-publishing.com  Thu Aug 23 20:41:27 2012
From: ben at fc-publishing.com (Ben Ringham)
Date: Thu, 23 Aug 2012 12:41:27 -0400
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 74, Issue 192
In-Reply-To: <mailman.17493.1345737084.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.17493.1345737084.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <CADk0+BdssJRrQ3zWwR9sau9P5+yQkpdPH5mJHrVi-j+_fA6P8A@mail.gmail.com>

When I installed fusion it was set this way. Under the extension edit menu
the "User Context" is set to "default". How, what and were need to change
to fix it?


>
>
>
> ---------- Forwarded message ----------
> From: Brian Foster <bdfoster at endigotech.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Thu, 23 Aug 2012 11:50:51 -0400
> Subject: Re: [Freeswitch-users] Unable to outbound with CounterPath eyebeam
> We've been through this before. Why are you on the public context with
> your internal extension? You should be on the default context. It's not
> doing anything because nothing is matching, and when that happens the call
> is hung up (with NORMAL CLEARNING).
>
> -BDF
>
> On Thu, Aug 23, 2012 at 10:56 AM, Ben Ringham <ben at fc-publishing.com>wrote:
>
>> Hello all,
>> Maybe you can help with an issue I have had from about day 2 of my
>> fusion/freeswitch install.
>>
>> I am able to receive calls inbound to my softphone (counterpath eyebeam
>> V1.5.19.4) When I try to dial out, or any extensions I am unable to
>> connect. Watching the CLI it the softphone seems to connect and talk with
>> freeswitch, then hangs up the call. Below is CLI of the call I am
>> attempting. Any input to correct this issue will greatly help.
>> Ben
>>
>> +OK log level  [7]
>> freeswitch at internal> 2012-08-23 10:45:42.332691 [DEBUG] sofia.c:6466 IP
>> 10.0.1.151 Approved by acl "domains[]". Access Granted.
>> 2012-08-23 10:45:42.333693 [NOTICE] switch_channel.c:812 New Channel
>> sofia/internal/5555 at 10.0.1.162 [c915f249-a2d1-4c5e-8900-8ae47f3deea1]
>> 2012-08-23 10:45:42.333693 [DEBUG] sofia.c:4744 Channel sofia/internal/
>> 5555 at 10.0.1.162 entering state [received][100]
>> 2012-08-23 10:45:42.333693 [DEBUG] sofia.c:4755 Remote SDP:
>> v=0
>> o=- 7 2 IN IP4 10.0.1.151
>> s=CounterPath eyeBeam 1.5
>> c=IN IP4 10.0.1.151
>> t=0 0
>> m=audio 51938 RTP/AVP 107 0 8 18 101
>> a=rtpmap:107 BV32/16000
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=yes
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=alt:1 3 : RQXbULH6 jeWr1bhp 10.0.1.151 51938
>> a=alt:2 2 : lIrXnsi1 yeNfshsQ 192.168.238.1 51938
>> a=alt:3 1 : np3FXIkK K342ICWN 192.168.64.1 51938
>>
>> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
>> [BV32:107:16000:20:0]/[G722:9:8000:20:64000]
>> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
>> [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
>> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
>> [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
>> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
>> [BV32:107:16000:20:0]/[GSM:3:8000:20:13200]
>> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
>> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>> 2012-08-23 10:45:42.334691 [DEBUG] sofia_glue.c:2760 Set Codec
>> sofia/internal/5555 at 10.0.1.162 PCMU/8000 20 ms 160 samples 64000 bits
>> 2012-08-23 10:45:42.334691 [DEBUG] switch_core_state_machine.c:320
>> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_NEW
>> 2012-08-23 10:45:42.334691 [DEBUG] switch_core_state_machine.c:338
>> (sofia/internal/5555 at 10.0.1.162) State NEW
>> 2012-08-23 10:45:42.334691 [DEBUG] sofia_glue.c:4733 Set 2833 dtmf
>> send/recv payload to 101
>> 2012-08-23 10:45:42.334691 [DEBUG] sofia.c:4922 (sofia/internal/
>> 5555 at 10.0.1.162) State Change CS_NEW -> CS_INIT
>> 2012-08-23 10:45:42.334691 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/internal/5555 at 10.0.1.162 [BREAK]
>> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:320
>> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_INIT
>> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:356
>> (sofia/internal/5555 at 10.0.1.162) State INIT
>> 2012-08-23 10:45:42.335694 [DEBUG] mod_sofia.c:84 sofia/internal/
>> 5555 at 10.0.1.162 SOFIA INIT
>> 2012-08-23 10:45:42.335694 [DEBUG] mod_sofia.c:124 (sofia/internal/
>> 5555 at 10.0.1.162) State Change CS_INIT -> CS_ROUTING
>> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/internal/5555 at 10.0.1.162 [BREAK]
>> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:356
>> (sofia/internal/5555 at 10.0.1.162) State INIT going to sleep
>> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:320
>> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_ROUTING
>> 2012-08-23 10:45:42.335694 [DEBUG] switch_channel.c:1664 (sofia/internal/
>> 5555 at 10.0.1.162) Callstate Change DOWN -> RINGING
>> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:359
>> (sofia/internal/5555 at 10.0.1.162) State ROUTING
>> 2012-08-23 10:45:42.335694 [DEBUG] mod_sofia.c:147 sofia/internal/
>> 5555 at 10.0.1.162 SOFIA ROUTING
>> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:77
>> sofia/internal/5555 at 10.0.1.162 Standard ROUTING
>> 2012-08-23 10:45:42.335694 [INFO] mod_dialplan_xml.c:331 Processing Ben
>> <5555>->18002267611 in context public
>> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->unloop]
>> continue=false
>> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (PASS) [unloop]
>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL) [unloop]
>> ${sip_looped_call}() =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->outside_call]
>> continue=true
>> Dialplan: sofia/internal/5555 at 10.0.1.162 Absolute Condition
>> [outside_call]
>> Dialplan: sofia/internal/5555 at 10.0.1.162 Action set(outside_call=true)
>> Dialplan: sofia/internal/5555 at 10.0.1.162 Action
>> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->call_debug]
>> continue=true
>> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL) [call_debug]
>> ${call_debug}(false) =~ /^true$/ break=never
>> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing
>> [public->public_extensions] continue=false
>> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL)
>> [public_extensions] destination_number(18002267611) =~ /^(10[01][0-9])$/
>> break=on-false
>> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->inbound to ext
>> 55] continue=false
>> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (PASS) [inbound to ext
>> 55] context(public) =~ /public/ break=on-false
>> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL) [inbound to ext
>> 55] destination_number(18002267611) =~ /4049371797/ break=on-false
>> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:119
>> (sofia/internal/5555 at 10.0.1.162) State Change CS_ROUTING -> CS_EXECUTE
>> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/internal/5555 at 10.0.1.162 [BREAK]
>> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:359
>> (sofia/internal/5555 at 10.0.1.162) State ROUTING going to sleep
>> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:320
>> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_EXECUTE
>> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:366
>> (sofia/internal/5555 at 10.0.1.162) State EXECUTE
>> 2012-08-23 10:45:42.337693 [DEBUG] mod_sofia.c:240 sofia/internal/
>> 5555 at 10.0.1.162 SOFIA EXECUTE
>> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:157
>> sofia/internal/5555 at 10.0.1.162 Standard EXECUTE
>> EXECUTE sofia/internal/5555 at 10.0.1.162 set(outside_call=true)
>> 2012-08-23 10:45:42.337693 [DEBUG] mod_dptools.c:1059 sofia/internal/
>> 5555 at 10.0.1.162 SET [outside_call]=[true]
>> EXECUTE sofia/internal/5555 at 10.0.1.162 set(RFC2822_DATE=Thu, 23 Aug 2012
>> 10:45:42 +0530)
>> 2012-08-23 10:45:42.338693 [DEBUG] mod_dptools.c:1059 sofia/internal/
>> 5555 at 10.0.1.162 SET [RFC2822_DATE]=[Thu, 23 Aug 2012 10:45:42 +0530]
>> 2012-08-23 10:45:42.338693 [NOTICE] switch_core_state_machine.c:189
>> sofia/internal/5555 at 10.0.1.162 has executed the last dialplan
>> instruction, hanging up.
>> 2012-08-23 10:45:42.338693 [DEBUG] switch_channel.c:2559 (sofia/internal/
>> 5555 at 10.0.1.162) Callstate Change RINGING -> HANGUP
>> 2012-08-23 10:45:42.338693 [NOTICE] switch_core_state_machine.c:191
>> Hangup sofia/internal/5555 at 10.0.1.162 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2012-08-23 10:45:42.338693 [DEBUG] switch_channel.c:2575 Send signal
>> sofia/internal/5555 at 10.0.1.162 [KILL]
>> 2012-08-23 10:45:42.338693 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/internal/5555 at 10.0.1.162 [BREAK]
>> 2012-08-23 10:45:42.338693 [DEBUG] switch_core_state_machine.c:366
>> (sofia/internal/5555 at 10.0.1.162) State EXECUTE going to sleep
>> 2012-08-23 10:45:42.338693 [DEBUG] switch_core_state_machine.c:320
>> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_HANGUP
>> 2012-08-23 10:45:42.339694 [DEBUG] switch_core_state_machine.c:560
>> (sofia/internal/5555 at 10.0.1.162) State HANGUP
>> 2012-08-23 10:45:42.339694 [DEBUG] mod_sofia.c:457 Channel sofia/internal/
>> 5555 at 10.0.1.162 hanging up, cause: NORMAL_CLEARING
>> 2012-08-23 10:45:42.339694 [DEBUG] mod_sofia.c:519 Responding to INVITE
>> with: 480
>> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:46
>> sofia/internal/5555 at 10.0.1.162 Standard HANGUP, cause: NORMAL_CLEARING
>> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:560
>> (sofia/internal/5555 at 10.0.1.162) State HANGUP going to sleep
>> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:351
>> (sofia/internal/5555 at 10.0.1.162) State Change CS_HANGUP -> CS_REPORTING
>> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/internal/5555 at 10.0.1.162 [BREAK]
>> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:320
>> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_REPORTING
>> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:620
>> (sofia/internal/5555 at 10.0.1.162) State REPORTING
>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:53
>> sofia/internal/5555 at 10.0.1.162 Standard REPORTING, cause: NORMAL_CLEARING
>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:620
>> (sofia/internal/5555 at 10.0.1.162) State REPORTING going to sleep
>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:345
>> (sofia/internal/5555 at 10.0.1.162) State Change CS_REPORTING -> CS_DESTROY
>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_session.c:1116 Send signal
>> sofia/internal/5555 at 10.0.1.162 [BREAK]
>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_session.c:1288 Session 22
>> (sofia/internal/5555 at 10.0.1.162) Locked, Waiting on external entities
>> 2012-08-23 10:45:42.489418 [NOTICE] switch_core_session.c:1306 Session 22
>> (sofia/internal/5555 at 10.0.1.162) Ended
>> 2012-08-23 10:45:42.489418 [NOTICE] switch_core_session.c:1308 Close
>> Channel sofia/internal/5555 at 10.0.1.162 [CS_DESTROY]
>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:449
>> (sofia/internal/5555 at 10.0.1.162) Callstate Change HANGUP -> DOWN
>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:452
>> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_DESTROY
>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:462
>> (sofia/internal/5555 at 10.0.1.162) State DESTROY
>> 2012-08-23 10:45:42.491481 [DEBUG] mod_sofia.c:362 sofia/internal/
>> 5555 at 10.0.1.162 SOFIA DESTROY
>> 2012-08-23 10:45:42.491481 [DEBUG] switch_core_state_machine.c:60
>> sofia/internal/5555 at 10.0.1.162 Standard DESTROY
>> 2012-08-23 10:45:42.491481 [DEBUG] switch_core_state_machine.c:462
>> (sofia/internal/5555 at 10.0.1.162) State DESTROY going to sleep
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
> _______________________________________________
> FreeSWITCH-users mailing 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 andrew at cassidywebservices.co.uk  Thu Aug 23 20:59:21 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 23 Aug 2012 17:59:21 +0100
Subject: [Freeswitch-users] Shared Call Appearance and Transferring
 Calls on Cisco SPA504G
In-Reply-To: <0D1C698866F66045A6201FD0F59CAC90014678A921@EX.frontier.local>
References: <0D1C698866F66045A6201FD0F59CAC90014678A921@EX.frontier.local>
Message-ID: <CAMbW9oLKbYPi5P_twkad_Ht+FrHWnEV5puB-ZuO8hzg_o=X1rA@mail.gmail.com>

I have an SPA303, and the process to transfer (either attended of blind) is
to press a key labeled 'xfer' or 'bxfer' then dial the line to transfer to.
Aside from that I've not played with SLA on these devices.

You can emulate these features using just one line per phone setting up the
others as speed dial, blf and call pickup using the syntax
fnc=sd+blf+cp;ext=<exten>

Make sure you set type to RFC3265_4235 on the Attendant keys tab, and add
the relevant dialplan entries to handle the interception of the calls (call
pickup prepends ** to the number)

Alternatively there's always call parking, too. You could set up 3 slot
parking lot, handle the hold event to park the call in a specific slot,
then getting the call back again is just a speed dial key.

That setup works for me, and is only a suggestion. As I've only got the one
device I can't offer any solutions for using SLA at this time.

On 23 August 2012 15:33, Colin Mason <cmason at frontiernetworks.ca> wrote:

> So I have been experimenting with a key system for an office with 3 Cisco
> SPA504G phones and FreeSWITCH 1.2.1. They require 3 shared lines. I have
> inbound and outbound calls working properly with the shared lines and the
> phones are configured properly with shared lines, Broadcom etc. Visually
> everything seems to work with the line notifications on the 3 phones.****
>
> ** **
>
> My internal profile has:****
>
> ** **
>
>     <param name="manage-shared-appearance" value="true"/>****
>
>     <param name="manage-presence" value="true"/>****
>
>     <param name="dbname" value="share_presence"/>****
>
>     <param name="presence-hosts" value="$${domain}"/>****
>
>     <param name="multiple-registrations" value="true"/>****
>
>     <param name="force-register-domain" value="$${domain}"/>****
>
>     <param name="force-subscription-domain" value="$${domain}"/>****
>
>     <param name="force-register-db-domain" value="$${domain}"/>****
>
>     <param name="presence-probe-on-register" value="true"/>****
>
>     <param name="presence-privacy" value="$${presence_privacy}"/>****
>
>     <param name="send-presence-on-register" value="true"/>****
>
> ** **
>
> I am having problems with transferring calls. If I put a call on hold on
> phone 1 and press the line 1 button on phone 1, the call resumes just fine.
> But if I put a call on hold on phone 1 and try to resume the call on phone
> 2 by pressing the blinking red line 1 button, the phone tries to establish
> a new call to the user associated with line 1 instead of taking
> (transferring) the call from phone 1 to phone 2.****
>
> ** **
>
> I was wondering if anybody could help?****
>
> ** **
>
> Colin Mason****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From bdfoster at endigotech.com  Thu Aug 23 21:03:00 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 23 Aug 2012 13:03:00 -0400
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 74, Issue 192
In-Reply-To: <CADk0+BdssJRrQ3zWwR9sau9P5+yQkpdPH5mJHrVi-j+_fA6P8A@mail.gmail.com>
References: <mailman.17493.1345737084.13443.freeswitch-users@lists.freeswitch.org>
	<CADk0+BdssJRrQ3zWwR9sau9P5+yQkpdPH5mJHrVi-j+_fA6P8A@mail.gmail.com>
Message-ID: <CAKV-jFzCfw4RFfKbEaZ-6qamYqquJ7LkevYdeaRf+VLqGTRVAQ@mail.gmail.com>

I don't know enough about Fusion to tell you what you should do. FusionPBX
has changed quite a bit lately and I haven't kept track of it. Here's the
line that shows you it's in the public context:

2012-08-23 10:45:42.335694 [INFO] mod_dialplan_xml.c:331 Processing Ben
<5555>->18002267611 in context public

Running through the public context, and based on your conditions, no
actions end up being queued for execution, so the call is hung up.
Something else that I noticed is that the endpoint isn't registering, it's
auth'd via the ACL. I don't know if that's what you intended, but I would
assume that's part of your problem. Here's the line that shows the endpoint
is auth'd via ACL:

2012-08-23 10:45:42.332691 [DEBUG] sofia.c:6466 IP 10.0.1.151 Approved by
acl "domains[]". Access Granted.

Get your phone to register to the internal profile, and if the problem
still exists then Mark might be able to help you out. If you really want
your phones auth'd via ACL, I haven't had to mess with ACL enough to be
able to tell you what to do. Someone else might want to chime in.

- BDF


On Thu, Aug 23, 2012 at 12:41 PM, Ben Ringham <ben at fc-publishing.com> wrote:

> When I installed fusion it was set this way. Under the extension edit menu
> the "User Context" is set to "default". How, what and were need to change
> to fix it?
>
>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Brian Foster <bdfoster at endigotech.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Date: Thu, 23 Aug 2012 11:50:51 -0400
>> Subject: Re: [Freeswitch-users] Unable to outbound with CounterPath
>> eyebeam
>> We've been through this before. Why are you on the public context with
>> your internal extension? You should be on the default context. It's not
>> doing anything because nothing is matching, and when that happens the call
>> is hung up (with NORMAL CLEARNING).
>>
>> -BDF
>>
>> On Thu, Aug 23, 2012 at 10:56 AM, Ben Ringham <ben at fc-publishing.com>wrote:
>>
>>> Hello all,
>>> Maybe you can help with an issue I have had from about day 2 of my
>>> fusion/freeswitch install.
>>>
>>> I am able to receive calls inbound to my softphone (counterpath eyebeam
>>> V1.5.19.4) When I try to dial out, or any extensions I am unable to
>>> connect. Watching the CLI it the softphone seems to connect and talk with
>>> freeswitch, then hangs up the call. Below is CLI of the call I am
>>> attempting. Any input to correct this issue will greatly help.
>>> Ben
>>>
>>> +OK log level  [7]
>>> freeswitch at internal> 2012-08-23 10:45:42.332691 [DEBUG] sofia.c:6466 IP
>>> 10.0.1.151 Approved by acl "domains[]". Access Granted.
>>> 2012-08-23 10:45:42.333693 [NOTICE] switch_channel.c:812 New Channel
>>> sofia/internal/5555 at 10.0.1.162 [c915f249-a2d1-4c5e-8900-8ae47f3deea1]
>>> 2012-08-23 10:45:42.333693 [DEBUG] sofia.c:4744 Channel sofia/internal/
>>> 5555 at 10.0.1.162 entering state [received][100]
>>> 2012-08-23 10:45:42.333693 [DEBUG] sofia.c:4755 Remote SDP:
>>> v=0
>>> o=- 7 2 IN IP4 10.0.1.151
>>> s=CounterPath eyeBeam 1.5
>>> c=IN IP4 10.0.1.151
>>> t=0 0
>>> m=audio 51938 RTP/AVP 107 0 8 18 101
>>> a=rtpmap:107 BV32/16000
>>> a=rtpmap:18 G729/8000
>>> a=fmtp:18 annexb=yes
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=alt:1 3 : RQXbULH6 jeWr1bhp 10.0.1.151 51938
>>> a=alt:2 2 : lIrXnsi1 yeNfshsQ 192.168.238.1 51938
>>> a=alt:3 1 : np3FXIkK K342ICWN 192.168.64.1 51938
>>>
>>> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
>>> [BV32:107:16000:20:0]/[G722:9:8000:20:64000]
>>> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
>>> [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000]
>>> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
>>> [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000]
>>> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
>>> [BV32:107:16000:20:0]/[GSM:3:8000:20:13200]
>>> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
>>> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
>>> 2012-08-23 10:45:42.333693 [DEBUG] sofia_glue.c:4619 Audio Codec Compare
>>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>> 2012-08-23 10:45:42.334691 [DEBUG] sofia_glue.c:2760 Set Codec
>>> sofia/internal/5555 at 10.0.1.162 PCMU/8000 20 ms 160 samples 64000 bits
>>> 2012-08-23 10:45:42.334691 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_NEW
>>> 2012-08-23 10:45:42.334691 [DEBUG] switch_core_state_machine.c:338
>>> (sofia/internal/5555 at 10.0.1.162) State NEW
>>> 2012-08-23 10:45:42.334691 [DEBUG] sofia_glue.c:4733 Set 2833 dtmf
>>> send/recv payload to 101
>>> 2012-08-23 10:45:42.334691 [DEBUG] sofia.c:4922 (sofia/internal/
>>> 5555 at 10.0.1.162) State Change CS_NEW -> CS_INIT
>>> 2012-08-23 10:45:42.334691 [DEBUG] switch_core_session.c:1116 Send
>>> signal sofia/internal/5555 at 10.0.1.162 [BREAK]
>>> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_INIT
>>> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:356
>>> (sofia/internal/5555 at 10.0.1.162) State INIT
>>> 2012-08-23 10:45:42.335694 [DEBUG] mod_sofia.c:84 sofia/internal/
>>> 5555 at 10.0.1.162 SOFIA INIT
>>> 2012-08-23 10:45:42.335694 [DEBUG] mod_sofia.c:124 (sofia/internal/
>>> 5555 at 10.0.1.162) State Change CS_INIT -> CS_ROUTING
>>> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_session.c:1116 Send
>>> signal sofia/internal/5555 at 10.0.1.162 [BREAK]
>>> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:356
>>> (sofia/internal/5555 at 10.0.1.162) State INIT going to sleep
>>> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_ROUTING
>>> 2012-08-23 10:45:42.335694 [DEBUG] switch_channel.c:1664 (sofia/internal/
>>> 5555 at 10.0.1.162) Callstate Change DOWN -> RINGING
>>> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:359
>>> (sofia/internal/5555 at 10.0.1.162) State ROUTING
>>> 2012-08-23 10:45:42.335694 [DEBUG] mod_sofia.c:147 sofia/internal/
>>> 5555 at 10.0.1.162 SOFIA ROUTING
>>> 2012-08-23 10:45:42.335694 [DEBUG] switch_core_state_machine.c:77
>>> sofia/internal/5555 at 10.0.1.162 Standard ROUTING
>>> 2012-08-23 10:45:42.335694 [INFO] mod_dialplan_xml.c:331 Processing Ben
>>> <5555>->18002267611 in context public
>>> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->unloop]
>>> continue=false
>>> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (PASS) [unloop]
>>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>>> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL) [unloop]
>>> ${sip_looped_call}() =~ /^true$/ break=on-false
>>> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->outside_call]
>>> continue=true
>>> Dialplan: sofia/internal/5555 at 10.0.1.162 Absolute Condition
>>> [outside_call]
>>> Dialplan: sofia/internal/5555 at 10.0.1.162 Action set(outside_call=true)
>>> Dialplan: sofia/internal/5555 at 10.0.1.162 Action
>>> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>>> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->call_debug]
>>> continue=true
>>> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL) [call_debug]
>>> ${call_debug}(false) =~ /^true$/ break=never
>>> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing
>>> [public->public_extensions] continue=false
>>> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL)
>>> [public_extensions] destination_number(18002267611) =~
>>> /^(10[01][0-9])$/ break=on-false
>>> Dialplan: sofia/internal/5555 at 10.0.1.162 parsing [public->inbound to
>>> ext 55] continue=false
>>> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (PASS) [inbound to ext
>>> 55] context(public) =~ /public/ break=on-false
>>> Dialplan: sofia/internal/5555 at 10.0.1.162 Regex (FAIL) [inbound to ext
>>> 55] destination_number(18002267611) =~ /4049371797/ break=on-false
>>> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:119
>>> (sofia/internal/5555 at 10.0.1.162) State Change CS_ROUTING -> CS_EXECUTE
>>> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_session.c:1116 Send
>>> signal sofia/internal/5555 at 10.0.1.162 [BREAK]
>>> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:359
>>> (sofia/internal/5555 at 10.0.1.162) State ROUTING going to sleep
>>> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_EXECUTE
>>> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:366
>>> (sofia/internal/5555 at 10.0.1.162) State EXECUTE
>>> 2012-08-23 10:45:42.337693 [DEBUG] mod_sofia.c:240 sofia/internal/
>>> 5555 at 10.0.1.162 SOFIA EXECUTE
>>> 2012-08-23 10:45:42.337693 [DEBUG] switch_core_state_machine.c:157
>>> sofia/internal/5555 at 10.0.1.162 Standard EXECUTE
>>> EXECUTE sofia/internal/5555 at 10.0.1.162 set(outside_call=true)
>>> 2012-08-23 10:45:42.337693 [DEBUG] mod_dptools.c:1059 sofia/internal/
>>> 5555 at 10.0.1.162 SET [outside_call]=[true]
>>> EXECUTE sofia/internal/5555 at 10.0.1.162 set(RFC2822_DATE=Thu, 23 Aug
>>> 2012 10:45:42 +0530)
>>> 2012-08-23 10:45:42.338693 [DEBUG] mod_dptools.c:1059 sofia/internal/
>>> 5555 at 10.0.1.162 SET [RFC2822_DATE]=[Thu, 23 Aug 2012 10:45:42 +0530]
>>> 2012-08-23 10:45:42.338693 [NOTICE] switch_core_state_machine.c:189
>>> sofia/internal/5555 at 10.0.1.162 has executed the last dialplan
>>> instruction, hanging up.
>>> 2012-08-23 10:45:42.338693 [DEBUG] switch_channel.c:2559 (sofia/internal/
>>> 5555 at 10.0.1.162) Callstate Change RINGING -> HANGUP
>>> 2012-08-23 10:45:42.338693 [NOTICE] switch_core_state_machine.c:191
>>> Hangup sofia/internal/5555 at 10.0.1.162 [CS_EXECUTE] [NORMAL_CLEARING]
>>> 2012-08-23 10:45:42.338693 [DEBUG] switch_channel.c:2575 Send signal
>>> sofia/internal/5555 at 10.0.1.162 [KILL]
>>> 2012-08-23 10:45:42.338693 [DEBUG] switch_core_session.c:1116 Send
>>> signal sofia/internal/5555 at 10.0.1.162 [BREAK]
>>> 2012-08-23 10:45:42.338693 [DEBUG] switch_core_state_machine.c:366
>>> (sofia/internal/5555 at 10.0.1.162) State EXECUTE going to sleep
>>> 2012-08-23 10:45:42.338693 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_HANGUP
>>> 2012-08-23 10:45:42.339694 [DEBUG] switch_core_state_machine.c:560
>>> (sofia/internal/5555 at 10.0.1.162) State HANGUP
>>> 2012-08-23 10:45:42.339694 [DEBUG] mod_sofia.c:457 Channel
>>> sofia/internal/5555 at 10.0.1.162 hanging up, cause: NORMAL_CLEARING
>>> 2012-08-23 10:45:42.339694 [DEBUG] mod_sofia.c:519 Responding to INVITE
>>> with: 480
>>> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:46
>>> sofia/internal/5555 at 10.0.1.162 Standard HANGUP, cause: NORMAL_CLEARING
>>> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:560
>>> (sofia/internal/5555 at 10.0.1.162) State HANGUP going to sleep
>>> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:351
>>> (sofia/internal/5555 at 10.0.1.162) State Change CS_HANGUP -> CS_REPORTING
>>> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_session.c:1116 Send
>>> signal sofia/internal/5555 at 10.0.1.162 [BREAK]
>>> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:320
>>> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_REPORTING
>>> 2012-08-23 10:45:42.340695 [DEBUG] switch_core_state_machine.c:620
>>> (sofia/internal/5555 at 10.0.1.162) State REPORTING
>>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:53
>>> sofia/internal/5555 at 10.0.1.162 Standard REPORTING, cause:
>>> NORMAL_CLEARING
>>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:620
>>> (sofia/internal/5555 at 10.0.1.162) State REPORTING going to sleep
>>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:345
>>> (sofia/internal/5555 at 10.0.1.162) State Change CS_REPORTING -> CS_DESTROY
>>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_session.c:1116 Send
>>> signal sofia/internal/5555 at 10.0.1.162 [BREAK]
>>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_session.c:1288 Session 22
>>> (sofia/internal/5555 at 10.0.1.162) Locked, Waiting on external entities
>>> 2012-08-23 10:45:42.489418 [NOTICE] switch_core_session.c:1306 Session
>>> 22 (sofia/internal/5555 at 10.0.1.162) Ended
>>> 2012-08-23 10:45:42.489418 [NOTICE] switch_core_session.c:1308 Close
>>> Channel sofia/internal/5555 at 10.0.1.162 [CS_DESTROY]
>>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:449
>>> (sofia/internal/5555 at 10.0.1.162) Callstate Change HANGUP -> DOWN
>>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:452
>>> (sofia/internal/5555 at 10.0.1.162) Running State Change CS_DESTROY
>>> 2012-08-23 10:45:42.489418 [DEBUG] switch_core_state_machine.c:462
>>> (sofia/internal/5555 at 10.0.1.162) State DESTROY
>>> 2012-08-23 10:45:42.491481 [DEBUG] mod_sofia.c:362 sofia/internal/
>>> 5555 at 10.0.1.162 SOFIA DESTROY
>>> 2012-08-23 10:45:42.491481 [DEBUG] switch_core_state_machine.c:60
>>> sofia/internal/5555 at 10.0.1.162 Standard DESTROY
>>> 2012-08-23 10:45:42.491481 [DEBUG] switch_core_state_machine.c:462
>>> (sofia/internal/5555 at 10.0.1.162) State DESTROY going to sleep
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>>
>> --
>> Brian D. Foster
>> Endigo Computer LLC
>> Email: bdfoster at endigotech.com
>> Phone: 317-800-7876
>> Indianapolis, Indiana, USA
>>
>> This message contains confidential information and is intended for those
>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>> you are not the intended recipient you are notified that disclosing,
>> copying, distributing or taking any action in reliance on the contents of
>> this information is strictly prohibited. E-mail transmission cannot be
>> guaranteed to be secure or error-free as information could be intercepted,
>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>> The sender therefore does not accept liability for any errors or omissions
>> in the contents of this message, which arise as a result of e-mail
>> transmission. If verification is required please request a hard-copy
>> version.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From lists at kavun.ch  Thu Aug 23 21:03:57 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 23 Aug 2012 13:03:57 -0400
Subject: [Freeswitch-users] Conflicting NAT handling on same user
	registering on different SIP profiles
Message-ID: <7FA642E1-C70C-4A5E-84E6-FD07B0CCD0CB@kavun.ch>

Hi all,

I have user 20 at domain registering both via SIP profile a and SIP profile b.
SIP profile A and B each have their own public IP.
On SIP profile A, user 20 at domain registers with a Softphone.
On SIP profile B, user 20 at domain registers with a Polycom phone.
Both the Softphone and the Polycom are behind the same NAT.

SIP profile A is using the default NAT configuration.
SIP profile B is using a special hack to handle NAT for Polycom phones, as follows:
<param name="NDLB-force-rport" value="true"/>

After that, if both SIP profile A and SIP profile B are loaded, only the Polycom is able to register. The softphone does gets by, packets go and come back, but FS does not seem to register it.
If I disable the Polycom profile, the Softphone can register fine.

My temporary solution is to register both under the Polycom profile, but it's a little unreliable.

What are your suggestions at this point?

Best,
Emrah

From anthony.minessale at gmail.com  Thu Aug 23 21:04:01 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 23 Aug 2012 12:04:01 -0500
Subject: [Freeswitch-users] RTP Timestamp changes after REFER - Question
 on RFC conformity
In-Reply-To: <CAAMQ8bReGJZdZeeiNeq5_38_Y=wrCd-8ARLzAgZe8kymyJ-gzQ@mail.gmail.com>
References: <CAAMQ8bRGeZm-7NWbRg4yoRZBqYTHEkXUCKUJJYgsEGZPZYNVgQ@mail.gmail.com>
	<CAKbxfG-6jiiH9HncmpAuPsknjmzgJntG_6eThBE6oYwckWtJ5Q@mail.gmail.com>
	<CAAMQ8bReGJZdZeeiNeq5_38_Y=wrCd-8ARLzAgZe8kymyJ-gzQ@mail.gmail.com>
Message-ID: <CAKbxfG-WX5Wqc00t=XC1S4nM55EmYLyeqqAN+XDx-mvztKdcGg@mail.gmail.com>

The problem here is you can easily blur the lines on what is correct
or incorrect and a possible fix might even break other things.

http://sidious.freeswitch.org/patches/rtp_reset.diff

Try this patch and see if its any better.  It will send a mark and new
SSRC every time a channel is in a new bridge situation.





On Thu, Aug 23, 2012 at 10:31 AM, Christian Benke <benkokakao at gmail.com> wrote:
> Hi!
>
> Here's a trace: http://poab.org/misc/one_way_audio_internal_g711.pcap
> (or http://filedump.org/files/yMnIRQUC1345735448.html for the archive)
>
> The REFER happens at Paket 1305, Time 25.662729
>
> The RTP-Paket that are sent to the C-endpoint after the REFER are
> continuous from A's point of view(As it is the continuation of the
> A-B-stream), but not from C's point of view.
>
> Regards,
> Christian
>
> On 23 August 2012 17:18, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> Do you have a pcap of it not setting the mark bit when the timestamp
>> changes because the code will always set the mark bit when the packet
>> is about to send is not exactly the next packet it expected to send
>> after the previous one/
>>
>>
>> On Thu, Aug 23, 2012 at 10:04 AM, Christian Benke <benkokakao at gmail.com> wrote:
>>> Hi!
>>>
>>> I had a problem with VOP Softclient(www.voiceoperatorpanel.com)  where
>>> the callee had one-way-audio after a call was REFERred to him. After
>>> lots of tracing, debugging and hairpulling i realized the problem lies
>>> in the changing rtp-timestamp when the RTP-stream is switched from the
>>> middleman to the initial call after the REFER. The Softclient was not
>>> able to cope with the changed timestamp and ignored the incoming
>>> RTP-packets, leading to no audio for the callee.
>>>
>>> I was eventually able to solve the problem by activating
>>> rtp-rewrite-timestamps on the profile(Also added it to
>>> http://wiki.freeswitch.org/wiki/RTP_Issues#Voiceoperatorpanel_VOP).
>>>
>>> However, i would like to know if FreeSWITCH/Sofia is working according
>>> to the RFC and if the Softclient is to blame for the problem(So i can
>>> file a bugreport with them).
>>>
>>> In this thread, Brian West states that  it's ok to skip forward in
>>> timestamps as long as the marker-bit is set:
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-July/060333.html
>>> However, the Marker-bit is not set by FreeSWITCH when the REFER occurs.
>>>
>>> I didn't find this stated in http://tools.ietf.org/html/rfc3550(But
>>> "timestamp" is mentioned a lot, so i may have missed it) but there's
>>> this bug-report for Asterisk, where the exact same problem is
>>> described and eventually handled:
>>> https://issues.asterisk.org/view.php?id=17007
>>>
>>> Could someone with more insight please elaborate?
>>>
>>> Best regards,
>>> Christian
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

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

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

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


From andrew at cassidywebservices.co.uk  Thu Aug 23 21:05:24 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Thu, 23 Aug 2012 18:05:24 +0100
Subject: [Freeswitch-users] ipv6 - any fans?
In-Reply-To: <CAKDfjgeY0UGZqm33Zrb=XdhEcgb9paz7sJ+dmi8_1PYZi35RaA@mail.gmail.com>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in> <503226CE.7040307@hw.ac.uk>
	<50323D6A.3020708@anew.com.ve> <2526639.vT3cT5qGKG@virtex>
	<5033C5B1.3070509@hw.ac.uk> <5036220C.9080203@gmail.com>
	<5036243F.4060601@thewinelake.com>
	<CAKDfjgdVYJBS0_WVdsSHE7=PPwRYW_RCSb0Pdvx0Yfbn4qWy4A@mail.gmail.com>
	<CAKDfjgeY0UGZqm33Zrb=XdhEcgb9paz7sJ+dmi8_1PYZi35RaA@mail.gmail.com>
Message-ID: <CAMbW9oJVVA04NqrE0gFyN955DMbcXEmHdqjBgOF24v_845reVw@mail.gmail.com>

I'm all for IPv6, just get your firewalling right, depending on your OS it
doesn't always share rules automatically. Don't forget your machine will
usually have at least 3 addresses too, which can of course be bound and
firewalled individually.

But as the others have said, the main issue is support in existing kit.
Only a fraction of my devices support IPv6. It's good for some things and
not doing something just because no-one else is will never push the
progression on. The more demand we create for IPv6, the more likely ISPs
are to take it up.

On 23 August 2012 15:43, Kristian Kielhofner <kris at kriskinc.com> wrote:

> Sorry, I just woke up: "there end of the tunnel" = "their end of the
> tunnel".
>
> On Thu, Aug 23, 2012 at 10:42 AM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
> > Assuming you're in the US (or most other parts of the world besides
> > Asia); it won't work.  Here's why:
> >
> > Almost no last mile providers will deliver native IPv6 (currently).
> > Your customers would have to use a tunneling service like HE to even
> > get to you.  First there are hassles setting up the tunnel in the
> > first place, then there are issues with latency, packet loss, jitter,
> > etc using a tunnel service that has to haul all of your IPv6 in IPv4
> > traffic back to their gateway in a central location.  VoIP is just
> > about the worst service you could use via an IPv6 tunnel.
> >
> > You could try to offer your own IPv6 tunneling service but there's
> > still the issue of customers setting up there end of the tunnel.
> > Besides, encapsulation sucks.
> >
> > The IPv4 global routing table has at least 424,000 networks:
> >
> > http://bgp.potaroo.net/index-bgp.html
> >
> > The IPv6 global routing table has 10,215 networks:
> >
> > http://bgp.potaroo.net/v6/as2.0/
> >
> > Granted IPv6 networks tend to be better organized and more "compact"
> > (at least as far as advertisements are concerned) but that's still a
> > factor of 40:1.
> >
> > On Thu, Aug 23, 2012 at 8:38 AM, Alex <alex at thewinelake.com> wrote:
> >> We've been wondering about trying to offer a service on ipv6 as it would
> >> get around various LAN issues.
> >>
> >> Just wondered what experiences (good or bad) people here have had with
> >> it and also why there isn't more "'buzz"?
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Kristian Kielhofner
>
>
>
> --
> Kristian Kielhofner
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From bdfoster at endigotech.com  Thu Aug 23 21:11:50 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 23 Aug 2012 13:11:50 -0400
Subject: [Freeswitch-users] FreeSWITCH Question
In-Reply-To: <CAKDfjgcnR4mTNfuqWckmsLOL_B6-p9fq8DzbiWpvSDgcCH4YEA@mail.gmail.com>
References: <5034019b.c456b60a.3fd2.ffffdc5cSMTPIN_ADDED@mx.google.com>
	<CALNZuwmN+xfGkW0HqqDpXdJ4WaW5o59BcX-RiuKdNo18QH5D7g@mail.gmail.com>
	<CD9593DE-1F02-4135-AEDD-7ED7C6A5FB22@visionutveckling.se>
	<63B00DD1DA6A364E9F64A3A0BD2FE7B612FB043F@BLUPRD0711MB413.namprd07.prod.outlook.com>
	<CAKDfjgcnR4mTNfuqWckmsLOL_B6-p9fq8DzbiWpvSDgcCH4YEA@mail.gmail.com>
Message-ID: <CAKV-jFyYmj6amU-vEcb_gZSpPMNBAp6EUh=k5f+oGJvdaTKesA@mail.gmail.com>

+1 here as well. 9000 calls is quite a bit, and at 90 cps that's going to
require some pretty serious DB usage. I'd be planning on database servers
to offload some of the work.

I also wouldn't risk 9000 calls if that server decides to fail. I know it's
FreeSWITCH, but the problems aren't limited to the software. Hardware
issues such as a bad HDD or a power failure are out of your control. As
humans, we're the weakest link. Plan for that too.

- BDF

P.S.: Please use a more descriptive subject for your next thread. 99% of
the threads on the mailing list are FreeSWITCH questions :)

On Thu, Aug 23, 2012 at 11:02 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> "And, I am at a loss as to what business model has so many calls, and
> can't afford a $500 server. So, just buy a server and move on."
>
> Once again, +1.
>
> On Thu, Aug 23, 2012 at 9:57 AM, Michael Giagnocavo <mgg at giagnocavo.net>
> wrote:
> > Just playing along with the whole "we can't get real hardware" idea:
> >
> >  - Try a 32-bit build, this may decrease memory usage significantly.
> >  - You didn't mention how the memory is actually used (is that 85%
> physically mapped?). Each thread gets a 240K (I think) allocation by
> default. You might try reducing that, but you'll need to test it and make
> sure you don't start overflowing.
> >
> > As much as I disagree with the "ten-thousand-threads" model, FS isn't
> going to change it anytime soon (and hey, it's working). And, I am at a
> loss as to what business model has so many calls, and can't afford a $500
> server. So, just buy a server and move on.
> >
> > -Michael
> >
> > -------
> > It is not so much the cps but the concurrent calls.
> >
> > I am looking at maintaining a 30 cps with a 5 min. avg. call length
> which results in a sustained 9000 concurrent calls.
> >
> > This results in 85% mem. usage on a 4 GB machine, However CPU is at 25%
> (total) on a dual core with hyper threading.
> >
> > Current hardware does not support increase memory.
> >
> > I am striving for about 75% cpu (appox. 90 cps).
> >
> > Jason
> >
> >
> > From: 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 Avi Marcus
> > Sent: Tuesday, August 21, 2012 2:58 PM
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] FreeSWITCH Question
> >
> > Do you have such a large cps/concurrent volume that you're actually
> seeing a performance hit?
> >
> > -Avi
> >
> > On Wed, Aug 22, 2012 at 12:35 AM, Jason Caulfield <
> jason.caulfield at intermetro.net<mailto:jason.caulfield at intermetro.net>>
> wrote:
> > I am using FreeSWITCH in media bypass mode to decrease memory and cpu
> usage.
> >
> > I notice that mod_sofia creates threads for each call leg to maintain
> session context.
> >
> > Do you know of a way to configure FreeSWITCH to use a table to maintain
> the context when in media bypass mode to reduce the number of threads?
> >
> > I am hoping that this will speed things up by reducing thread context
> switching and reduce memory usage by decreasing memory allocation for each
> thread.
> >
> > Thanks for the help,
> > Jason
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> > http://www.freeswitchsolutions.com
> >
> > 
> 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > 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:50340a7d32766392456629!
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> > http://www.freeswitchsolutions.com
> >
> > 
> 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > 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:50340a7d32766392456629!
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> --
> Kristian Kielhofner
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
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From bdfoster at endigotech.com  Thu Aug 23 21:17:37 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 23 Aug 2012 13:17:37 -0400
Subject: [Freeswitch-users] ipv6 - any fans?
In-Reply-To: <CAMbW9oJVVA04NqrE0gFyN955DMbcXEmHdqjBgOF24v_845reVw@mail.gmail.com>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in> <503226CE.7040307@hw.ac.uk>
	<50323D6A.3020708@anew.com.ve> <2526639.vT3cT5qGKG@virtex>
	<5033C5B1.3070509@hw.ac.uk> <5036220C.9080203@gmail.com>
	<5036243F.4060601@thewinelake.com>
	<CAKDfjgdVYJBS0_WVdsSHE7=PPwRYW_RCSb0Pdvx0Yfbn4qWy4A@mail.gmail.com>
	<CAKDfjgeY0UGZqm33Zrb=XdhEcgb9paz7sJ+dmi8_1PYZi35RaA@mail.gmail.com>
	<CAMbW9oJVVA04NqrE0gFyN955DMbcXEmHdqjBgOF24v_845reVw@mail.gmail.com>
Message-ID: <CAKV-jFxngTc62tPp6hVKKEbc8kNhtiV6_pWabanNWLp54r4esA@mail.gmail.com>

+1. Most of the servers I have actually do have IPv6 addresses, however I
do not use them. I really just don't have the experience to deploy IPv6
safely. Those servers all have public IP's so really for me there isn't
much benefit (yet), so it's not on my priority list. FS ipv-6 profile is
immediately removed prior to first start of FreeSWITCH.

On Thu, Aug 23, 2012 at 1:05 PM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> I'm all for IPv6, just get your firewalling right, depending on your OS it
> doesn't always share rules automatically. Don't forget your machine will
> usually have at least 3 addresses too, which can of course be bound and
> firewalled individually.
>
> But as the others have said, the main issue is support in existing kit.
> Only a fraction of my devices support IPv6. It's good for some things and
> not doing something just because no-one else is will never push the
> progression on. The more demand we create for IPv6, the more likely ISPs
> are to take it up.
>
> On 23 August 2012 15:43, Kristian Kielhofner <kris at kriskinc.com> wrote:
>
>> Sorry, I just woke up: "there end of the tunnel" = "their end of the
>> tunnel".
>>
>> On Thu, Aug 23, 2012 at 10:42 AM, Kristian Kielhofner <kris at kriskinc.com>
>> wrote:
>> > Assuming you're in the US (or most other parts of the world besides
>> > Asia); it won't work.  Here's why:
>> >
>> > Almost no last mile providers will deliver native IPv6 (currently).
>> > Your customers would have to use a tunneling service like HE to even
>> > get to you.  First there are hassles setting up the tunnel in the
>> > first place, then there are issues with latency, packet loss, jitter,
>> > etc using a tunnel service that has to haul all of your IPv6 in IPv4
>> > traffic back to their gateway in a central location.  VoIP is just
>> > about the worst service you could use via an IPv6 tunnel.
>> >
>> > You could try to offer your own IPv6 tunneling service but there's
>> > still the issue of customers setting up there end of the tunnel.
>> > Besides, encapsulation sucks.
>> >
>> > The IPv4 global routing table has at least 424,000 networks:
>> >
>> > http://bgp.potaroo.net/index-bgp.html
>> >
>> > The IPv6 global routing table has 10,215 networks:
>> >
>> > http://bgp.potaroo.net/v6/as2.0/
>> >
>> > Granted IPv6 networks tend to be better organized and more "compact"
>> > (at least as far as advertisements are concerned) but that's still a
>> > factor of 40:1.
>> >
>> > On Thu, Aug 23, 2012 at 8:38 AM, Alex <alex at thewinelake.com> wrote:
>> >> We've been wondering about trying to offer a service on ipv6 as it
>> would
>> >> get around various LAN issues.
>> >>
>> >> Just wondered what experiences (good or bad) people here have had with
>> >> it and also why there isn't more "'buzz"?
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Kristian Kielhofner
>>
>>
>>
>> --
>> Kristian Kielhofner
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
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From peter.olsson at visionutveckling.se  Thu Aug 23 21:17:14 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 23 Aug 2012 17:17:14 +0000
Subject: [Freeswitch-users] RTP Timestamp changes after REFER - Question
 on RFC conformity
Message-ID: <1FFF97C269757C458224B7C895F35F1514CFC3@cantor.std.visionutv.se>

Also, in current git head it might be possible that the RTP bug CHANGE_SSRC_ON_MARKER helps. It will change to a new SSRC every time the marker is sent. However, I didn't look into the pcaps, so it's possible that it's not enough for this specific case.

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 23 augusti 2012 19:04
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] RTP Timestamp changes after REFER - Question on RFC conformity

The problem here is you can easily blur the lines on what is correct or incorrect and a possible fix might even break other things.

http://sidious.freeswitch.org/patches/rtp_reset.diff

Try this patch and see if its any better.  It will send a mark and new SSRC every time a channel is in a new bridge situation.





On Thu, Aug 23, 2012 at 10:31 AM, Christian Benke <benkokakao at gmail.com> wrote:
> Hi!
>
> Here's a trace: http://poab.org/misc/one_way_audio_internal_g711.pcap
> (or http://filedump.org/files/yMnIRQUC1345735448.html for the archive)
>
> The REFER happens at Paket 1305, Time 25.662729
>
> The RTP-Paket that are sent to the C-endpoint after the REFER are 
> continuous from A's point of view(As it is the continuation of the 
> A-B-stream), but not from C's point of view.
>
> Regards,
> Christian
>
> On 23 August 2012 17:18, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> Do you have a pcap of it not setting the mark bit when the timestamp 
>> changes because the code will always set the mark bit when the packet 
>> is about to send is not exactly the next packet it expected to send 
>> after the previous one/
>>
>>
>> On Thu, Aug 23, 2012 at 10:04 AM, Christian Benke <benkokakao at gmail.com> wrote:
>>> Hi!
>>>
>>> I had a problem with VOP Softclient(www.voiceoperatorpanel.com)  
>>> where the callee had one-way-audio after a call was REFERred to him. 
>>> After lots of tracing, debugging and hairpulling i realized the 
>>> problem lies in the changing rtp-timestamp when the RTP-stream is 
>>> switched from the middleman to the initial call after the REFER. The 
>>> Softclient was not able to cope with the changed timestamp and 
>>> ignored the incoming RTP-packets, leading to no audio for the callee.
>>>
>>> I was eventually able to solve the problem by activating 
>>> rtp-rewrite-timestamps on the profile(Also added it to 
>>> http://wiki.freeswitch.org/wiki/RTP_Issues#Voiceoperatorpanel_VOP).
>>>
>>> However, i would like to know if FreeSWITCH/Sofia is working 
>>> according to the RFC and if the Softclient is to blame for the 
>>> problem(So i can file a bugreport with them).
>>>
>>> In this thread, Brian West states that  it's ok to skip forward in 
>>> timestamps as long as the marker-bit is set:
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-July/060
>>> 333.html However, the Marker-bit is not set by FreeSWITCH when the 
>>> REFER occurs.
>>>
>>> I didn't find this stated in http://tools.ietf.org/html/rfc3550(But
>>> "timestamp" is mentioned a lot, so i may have missed it) but there's 
>>> this bug-report for Asterisk, where the exact same problem is 
>>> described and eventually handled:
>>> https://issues.asterisk.org/view.php?id=17007
>>>
>>> Could someone with more insight please elaborate?
>>>
>>> Best regards,
>>> Christian
>>>
>>> ____________________________________________________________________
>>> _____ Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>>  
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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-u
>>> sers
>>> 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
>>
>> _____________________________________________________________________
>> ____ Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>  
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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-us
>> ers
>> http://www.freeswitch.org
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-use
> rs
> http://www.freeswitch.org



--
Anthony Minessale II

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

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

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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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

!DSPAM:5036605732761203987688!



From adam.kelloway at newpace.ca  Thu Aug 23 21:54:54 2012
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Thu, 23 Aug 2012 14:54:54 -0300
Subject: [Freeswitch-users] sip-router or opensips for good
 interoperability with freeswitch?
In-Reply-To: <1345739304.23436.89.camel@luna.madrid.commsmundi.com>
References: <1345739304.23436.89.camel@luna.madrid.commsmundi.com>
Message-ID: <50366E6E.4050902@newpace.ca>

The are both pretty well functionally the same. You might find that 
Kamailo's .cfg file is easier to work with, as it allows you to name the 
routes (among other things), and it has a decent default cfg to use as a 
starting point. I have known people to have success with both opensips 
and kamailio.

On 23/08/2012 1:28 PM, Fran?ois Delawarde wrote:
> Hi,
>
> Which of opensips or sip-router (kamailio) do you guys recommend to use
> in front of freeswitch to create some big reliable cluster?
>
> I'm not sure if there is any huge difference as both seem to share a
> same origin. Should I just throw a dice?, or is there one that is
> clearly better than the other (or one I should avoid) in terms of
> reliability / performance, or friendliness with FS?
>
> Thanks,

-- 
Adam
-- 
NewPace Logo 	
	
	
	Adam Kelloway

	Software Engineer, NewPace
phone 	+1 (902) 406--8375  x1031
email 	Adam.Kelloway at NewPace.com <mailto:Adam.Kelloway at newpace.com>
aim <aim:GoIm?screenname=Adam.Kelloway at newpace.com>/msn 
<msnim:chat?contact=Adam.Kelloway at newpace.com> 	Adam.Kelloway 
<aim:GoIm?screenname=Adam.Kelloway at newpace.ca>@NewPace.ca

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From curriegrad2004 at gmail.com  Thu Aug 23 22:10:12 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 23 Aug 2012 11:10:12 -0700
Subject: [Freeswitch-users] ipv6 - any fans?
In-Reply-To: <CAKV-jFxngTc62tPp6hVKKEbc8kNhtiV6_pWabanNWLp54r4esA@mail.gmail.com>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in> <503226CE.7040307@hw.ac.uk>
	<50323D6A.3020708@anew.com.ve> <2526639.vT3cT5qGKG@virtex>
	<5033C5B1.3070509@hw.ac.uk> <5036220C.9080203@gmail.com>
	<5036243F.4060601@thewinelake.com>
	<CAKDfjgdVYJBS0_WVdsSHE7=PPwRYW_RCSb0Pdvx0Yfbn4qWy4A@mail.gmail.com>
	<CAKDfjgeY0UGZqm33Zrb=XdhEcgb9paz7sJ+dmi8_1PYZi35RaA@mail.gmail.com>
	<CAMbW9oJVVA04NqrE0gFyN955DMbcXEmHdqjBgOF24v_845reVw@mail.gmail.com>
	<CAKV-jFxngTc62tPp6hVKKEbc8kNhtiV6_pWabanNWLp54r4esA@mail.gmail.com>
Message-ID: <CAMWiMy9gq8iZZqFbG6OZBiXa-p+9mAJgVSy7LrX-kJcf1g9Hew@mail.gmail.com>

IPv6 isn't hard to deploy. In fact it's way easier to deploy than IPv4. I
would agree that getting the firewalling right is the crucial part on
getting IPv6 deployed securely. A perfect example for firewalling an IPv6
network with iptables via the FORWARD chain can be done this way:

-A FORWARD -s [Your IPv6 Subnet Here] -j ACCEPT
-A FORWARD -d [Your IPv6 Subnet Here]  -m state --state RELATED,ESTABLISHED
-j ACCEPT
-A FORWARD -j REJECT

On Thu, Aug 23, 2012 at 10:17 AM, Brian Foster <bdfoster at endigotech.com>wrote:

> +1. Most of the servers I have actually do have IPv6 addresses, however I
> do not use them. I really just don't have the experience to deploy IPv6
> safely. Those servers all have public IP's so really for me there isn't
> much benefit (yet), so it's not on my priority list. FS ipv-6 profile is
> immediately removed prior to first start of FreeSWITCH.
>
>
> On Thu, Aug 23, 2012 at 1:05 PM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
>> I'm all for IPv6, just get your firewalling right, depending on your OS
>> it doesn't always share rules automatically. Don't forget your machine will
>> usually have at least 3 addresses too, which can of course be bound and
>> firewalled individually.
>>
>> But as the others have said, the main issue is support in existing kit.
>> Only a fraction of my devices support IPv6. It's good for some things and
>> not doing something just because no-one else is will never push the
>> progression on. The more demand we create for IPv6, the more likely ISPs
>> are to take it up.
>>
>> On 23 August 2012 15:43, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>
>>> Sorry, I just woke up: "there end of the tunnel" = "their end of the
>>> tunnel".
>>>
>>> On Thu, Aug 23, 2012 at 10:42 AM, Kristian Kielhofner <kris at kriskinc.com>
>>> wrote:
>>> > Assuming you're in the US (or most other parts of the world besides
>>> > Asia); it won't work.  Here's why:
>>> >
>>> > Almost no last mile providers will deliver native IPv6 (currently).
>>> > Your customers would have to use a tunneling service like HE to even
>>> > get to you.  First there are hassles setting up the tunnel in the
>>> > first place, then there are issues with latency, packet loss, jitter,
>>> > etc using a tunnel service that has to haul all of your IPv6 in IPv4
>>> > traffic back to their gateway in a central location.  VoIP is just
>>> > about the worst service you could use via an IPv6 tunnel.
>>> >
>>> > You could try to offer your own IPv6 tunneling service but there's
>>> > still the issue of customers setting up there end of the tunnel.
>>> > Besides, encapsulation sucks.
>>> >
>>> > The IPv4 global routing table has at least 424,000 networks:
>>> >
>>> > http://bgp.potaroo.net/index-bgp.html
>>> >
>>> > The IPv6 global routing table has 10,215 networks:
>>> >
>>> > http://bgp.potaroo.net/v6/as2.0/
>>> >
>>> > Granted IPv6 networks tend to be better organized and more "compact"
>>> > (at least as far as advertisements are concerned) but that's still a
>>> > factor of 40:1.
>>> >
>>> > On Thu, Aug 23, 2012 at 8:38 AM, Alex <alex at thewinelake.com> wrote:
>>> >> We've been wondering about trying to offer a service on ipv6 as it
>>> would
>>> >> get around various LAN issues.
>>> >>
>>> >> Just wondered what experiences (good or bad) people here have had with
>>> >> it and also why there isn't more "'buzz"?
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> 
>>> >> 
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://wiki.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> > Kristian Kielhofner
>>>
>>>
>>>
>>> --
>>> Kristian Kielhofner
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>>
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>>
>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>  *03300 100 961
>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mike at jerris.com  Thu Aug 23 22:24:15 2012
From: mike at jerris.com (Michael Jerris)
Date: Thu, 23 Aug 2012 14:24:15 -0400
Subject: [Freeswitch-users] No voice heard
In-Reply-To: <CAPxXe8XFonrLKFkHQZ4kx0r4Z-8PHX2rqyq978CRDa+Ghf_PdA@mail.gmail.com>
References: <CAPxXe8XFonrLKFkHQZ4kx0r4Z-8PHX2rqyq978CRDa+Ghf_PdA@mail.gmail.com>
Message-ID: <779BE05F-F4BF-417D-9321-71E4B6B036D7@jerris.com>

mod_httapi allows you to use the http:// url directly without http_get function.  I suspect it is something with the format of the wav file.  Does the wav file work if its already on the machine and referenced directly?


On Aug 23, 2012, at 6:14 AM, Pete Kay <petedao at gmail.com> wrote:

> Hi,
> I tried the following command.
> <action application="playback"
> data="${http_get(http://myserver.yo/media/hello_world.wav)}"/>
> 
> After I replaced the url to a correct URL, file can be downloaded but
> no voice is heard.
> 
> Any idea how to troubleshoot this?



From cmason at frontiernetworks.ca  Thu Aug 23 22:46:04 2012
From: cmason at frontiernetworks.ca (Colin Mason)
Date: Thu, 23 Aug 2012 14:46:04 -0400
Subject: [Freeswitch-users] sip-router or opensips for good
 interoperability with freeswitch?
In-Reply-To: <50366E6E.4050902@newpace.ca>
References: <1345739304.23436.89.camel@luna.madrid.commsmundi.com>
	<50366E6E.4050902@newpace.ca>
Message-ID: <0D1C698866F66045A6201FD0F59CAC90014678A962@EX.frontier.local>

I use OpenSIPS to route SIP to and from 30 or so FreeSWITCH virtual machines. It works flawlessly. I would recommend using 1.7.2 instead of the 1.8 branch as I have found it is not completely stable yet.

Colin

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Kelloway
Sent: Thursday, August 23, 2012 1:55 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] sip-router or opensips for good interoperability with freeswitch?

The are both pretty well functionally the same. You might find that Kamailo's .cfg file is easier to work with, as it allows you to name the routes (among other things), and it has a decent default cfg to use as a starting point. I have known people to have success with both opensips and kamailio.
On 23/08/2012 1:28 PM, Fran?ois Delawarde wrote:

Hi,



Which of opensips or sip-router (kamailio) do you guys recommend to use

in front of freeswitch to create some big reliable cluster?



I'm not sure if there is any huge difference as both seem to share a

same origin. Should I just throw a dice?, or is there one that is

clearly better than the other (or one I should avoid) in terms of

reliability / performance, or friendliness with FS?



Thanks,

--
Adam
--
[cid:image001.png at 01CD813E.058CA030]

Adam Kelloway


Software Engineer, NewPace

phone

+1 (902) 406-8375  x1031

email

Adam.Kelloway at NewPace.com<mailto:Adam.Kelloway at newpace.com>

aim<aim:GoIm?screenname=Adam.Kelloway at newpace.com>/msn<msnim:chat?contact=Adam.Kelloway at newpace.com>

Adam.Kelloway<aim:GoIm?screenname=Adam.Kelloway at newpace.ca>@NewPace.ca


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From moises.silva at gmail.com  Fri Aug 24 01:51:59 2012
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 23 Aug 2012 17:51:59 -0400
Subject: [Freeswitch-users] FreeSWITCH and ISDN
In-Reply-To: <CAKzWOxWdra3sXuCZ_0KSn2GmUfmD85pNxiLWmT3xFGZYSC=5Fw@mail.gmail.com>
References: <CADnUQS_eejGVz3a47EX2jO9d5C7zf-o03HTuimi3Sbm=kPJrSw@mail.gmail.com>
	<CAKzWOxWdra3sXuCZ_0KSn2GmUfmD85pNxiLWmT3xFGZYSC=5Fw@mail.gmail.com>
Message-ID: <CAA4nhyDmjc8ERHeixhfEZdu__Qb+YxxJWyhwBvvcvV7VRosPMg@mail.gmail.com>

On Thu, Aug 23, 2012 at 12:04 PM, Michael Collins <msc at freeswitch.org>wrote:

> Nowadays the whole Digium vs. Sangoma thing has boiled down to preference
> more than anything else. They are both pretty reliable (I've used both) so
> I wouldn't use that metric as a tie-breaker. If money is the ultimate
> deciding factor then a Digium clone is probably the cheapest. However, if
> you want to support the company who has supported the FreeSWITCH community
> the most then it's hands down Sangoma. Also, we know all the Sangoma guys
> personally (Hi Moy!) and they have produced the bulk of mod_freetdm. Buying
> Sangoma is a great way to support FreeSWITCH.
>
>
Just to complement Michael's accurate description of the situation (Hi
Michael!) ...

There is a few other differences, such as the fact that if you ever need
SS7, there is no SS7 stack into FreeSWITCH for Digium boards, just for
Sangoma (due to SS7 stack restrictions).

For ISDN, although I know a few people using libpri in FreeTDM, the Sangoma
SS7 stack in FreeSWITCH (ftmod_sangoma_isdn) it is the one more actively
developed and under continuous improvement and professionally supported
(not just community support). If you go with Digium cards you have to stick
with the libpri module pretty much and the community support.

All in all, my guess (although probably biased) is that most FreeSWITCH
users use Sangoma cards and therefore if you run into issues you are more
likely to get help (either via community or via Sangoma support) if you use
a Sangoma card.

Cheers,

*Moises Silva
**Manager, Software Engineering***

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



**<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>

Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`|
| YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
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From jason at jasonjgw.net  Fri Aug 24 04:34:37 2012
From: jason at jasonjgw.net (Jason White)
Date: Fri, 24 Aug 2012 00:34:37 +0000 (UTC)
Subject: [Freeswitch-users] ipv6 - any fans?
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<CAKDfjgdVYJBS0_WVdsSHE7=PPwRYW_RCSb0Pdvx0Yfbn4qWy4A@mail.gmail.com>
	<CAKDfjgeY0UGZqm33Zrb=XdhEcgb9paz7sJ+dmi8_1PYZi35RaA@mail.gmail.com>
	<CAMbW9oJVVA04NqrE0gFyN955DMbcXEmHdqjBgOF24v_845reVw@mail.gmail.com>
Message-ID: <k16i6s$q2i$1@ger.gmane.org>

Andrew Cassidy  <freeswitch-users at lists.freeswitch.org> wrote:
>
>I'm all for IPv6, just get your firewalling right, depending on your OS it
>doesn't always share rules automatically. Don't forget your machine will
>usually have at least 3 addresses too, which can of course be bound and
>firewalled individually.

If you need a firewall, then you do indeed have to make sure that it's
configured appropriately for both IPv4 and IPv6. There are lazy/uninformed
administrators who think they can rely on NAT as a substitute for a firewall.

I have native IPv6 access via my ISP and FreeSWITCH works well with it. I have
also used FreeSWITCH over IPv6-in-IPv4 tunnels with surprisingly good
reliability, but in recent years with the native IPv6 this hasn't been
necessary except occasionally with a laptop on an IPv4-only network.

Deployment in a home office environment was easy enough, except initially for
the DHCPv6 prefix delegation and routing over the ppp connection to the ISP.




From gmaruzz at gmail.com  Fri Aug 24 05:31:31 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 24 Aug 2012 03:31:31 +0200
Subject: [Freeswitch-users] mod_gsmopen: ROAMING Network ERROR
In-Reply-To: <CAMjaK3AEPEc8BbTanipJ0s4dOO2qgg2xAkK6pYkaU8-bLSG8mg@mail.gmail.com>
References: <CAD=Wdqi_rbz3Z0XrYcS_CP29wSV94G8R7cJpHKRqZoeNorhmxw@mail.gmail.com>
	<CAMjaK3B-Y+nswvVnj49Jh_+sZCJKvcXiSmhJ7ksMWo6UXbGt9Q@mail.gmail.com>
	<CALXCt0q+CZsOBL6Uv0_W9TMje+qZGb+rguej-cgMqRNQA9mo6w@mail.gmail.com>
	<CAMjaK3AEPEc8BbTanipJ0s4dOO2qgg2xAkK6pYkaU8-bLSG8mg@mail.gmail.com>
Message-ID: <CALXCt0p_PKTxxgi1pgqN21GZdG+cPw_Sg+pXwfqcUT-BVOH5SA@mail.gmail.com>

Dimitry,
thanks for the nice words!

Please, do expand the wiki page adding all that you think can be
useful or interesting. Please. I will fix the English, if needed. No
worry.

I'll be back from holidays first week of Sept, ciao for now :)

-giovanni



On 8/23/12, Dmitry Lysenko <dvl36.ripe.nick at gmail.com> wrote:
> Hi Giovanni
>
> Thank you for good software. I used asterisk and chan_dongle  before, but
> freeswitch and gsmopen is leaving them behind.
> As for wiki, sorry but my english is not good enough.
> I have some questions about gsmopen functionality. But after your holidays.
>
> Thank you.
>
> 2012/8/21 Giovanni Maruzzelli <gmaruzz at gmail.com>
>
>> hi Sameer,
>> I'm in holidays.
>>
>> If problems persists, please contact me after Sept 10th.
>>
>> Dimitri, you're right. Btw, you seems very knowledgeable about gsmopen
>> and related stuff. Please feel free to modify expand mod-gsmopen wiki
>> page and/or send patches to jira. Thanks in advance Dimitri!
>>
>> -giovanni
>>
>> On 8/21/12, Dmitry Lysenko <dvl36.ripe.nick at gmail.com> wrote:
>> > Hello.
>> > I have the same "ROAMING Network" ALERTs, it's annoying, but everything
>> > working OK.
>> > Checked with  Huawei E1550.
>> >
>> > P.S. Are you sure that your Huawei E169  is voice enabled?
>> >
>> > 2012/8/21 Josep Maria Vi?olas <vinolasbcn at gmail.com>
>> >
>> >> I have a Huawei E169 with a SIM from a "virtual provider" that uses
>> other
>> >> provider's networks, so it is allways technically 'roaming'.
>> >>
>> >> With mod_gsmopen the problem is that throws an error about "ROAMING
>> >> Network" and also an ALERT. I don't know if that is why it is not able
>> to
>> >> make calls. In that case, how can I tell the module that roaming is
>> >> not
>> a
>> >> bad thing??
>> >>
>> >> This is freeswitch debug output when I load mod_gsmmobile:
>> >> ------------------------------------------
>> >>
>> > ...
>> >
>> >> ------------------------------------------
>> >> ** From extension 1000 I hear a calling tone even the Freeswitch log
>> says
>> >> that ATD+N? command failed.
>> >>
>> >> Anyone knows if the problem is the ROAMING Network error??
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 curriegrad2004 at gmail.com  Fri Aug 24 07:04:17 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 23 Aug 2012 20:04:17 -0700
Subject: [Freeswitch-users] ipv6 - any fans?
In-Reply-To: <k16i6s$q2i$1@ger.gmane.org>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<CAKDfjgdVYJBS0_WVdsSHE7=PPwRYW_RCSb0Pdvx0Yfbn4qWy4A@mail.gmail.com>
	<CAKDfjgeY0UGZqm33Zrb=XdhEcgb9paz7sJ+dmi8_1PYZi35RaA@mail.gmail.com>
	<CAMbW9oJVVA04NqrE0gFyN955DMbcXEmHdqjBgOF24v_845reVw@mail.gmail.com>
	<k16i6s$q2i$1@ger.gmane.org>
Message-ID: <CAMWiMy-+GZfa50FUxgKWaKwKzGRiqjLgQf83eYL2rT25+40cFw@mail.gmail.com>

Sadly a lot of places imply that Nat provides some form of security when it
really wasn't meant for that usage.
On Aug 23, 2012 5:36 PM, "Jason White" <jason at jasonjgw.net> wrote:

> Andrew Cassidy  <freeswitch-users at lists.freeswitch.org> wrote:
> >
> >I'm all for IPv6, just get your firewalling right, depending on your OS it
> >doesn't always share rules automatically. Don't forget your machine will
> >usually have at least 3 addresses too, which can of course be bound and
> >firewalled individually.
>
> If you need a firewall, then you do indeed have to make sure that it's
> configured appropriately for both IPv4 and IPv6. There are lazy/uninformed
> administrators who think they can rely on NAT as a substitute for a
> firewall.
>
> I have native IPv6 access via my ISP and FreeSWITCH works well with it. I
> have
> also used FreeSWITCH over IPv6-in-IPv4 tunnels with surprisingly good
> reliability, but in recent years with the native IPv6 this hasn't been
> necessary except occasionally with a laptop on an IPv4-only network.
>
> Deployment in a home office environment was easy enough, except initially
> for
> the DHCPv6 prefix delegation and routing over the ppp connection to the
> ISP.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lists at kavun.ch  Fri Aug 24 07:34:52 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 23 Aug 2012 23:34:52 -0400
Subject: [Freeswitch-users] Dial out within a conference with early media
In-Reply-To: <CAKzWOxUnmh8LW=YAHvkJ8GoHP+mG0xs2vkhuiuguL0rt9iBCug@mail.gmail.com>
References: <EC5B3D9E-68E3-4974-AA17-EFE65D96CC52@kavun.ch>
	<CAKzWOxUnmh8LW=YAHvkJ8GoHP+mG0xs2vkhuiuguL0rt9iBCug@mail.gmail.com>
Message-ID: <996F119D-E4CD-4A55-BA4C-FCBA8CE27426@kavun.ch>

Hi Michael,

I will do some more testing and get back to you with the results.
I do not ignore Early Media and the info message is a carrier audio message played in progress mode, not a SIP INFO. Sorry for the confusion.

Thanks,
Emrah
On Aug 23, 2012, at 11:38 AM, Michael Collins <msc at freeswitch.org> wrote:

> For item #5 - is that a SIP INFO message or is that some audio saying that the user is not available? If it's early media it should cause the dial to be "successful" unless you're explicitly setting ignore_early_media=true in your dialstring.
> 
> -MC
> 
> On Wed, Aug 22, 2012 at 5:20 PM, Emrah <lists at kavun.ch> wrote:
> Hi guys,
> 
> Is there an option to dial with early media support from within a conference?
> 
> This is what happens now:
> 
> 1. I have an active conf. 
> 2. I decide to pull in an additional user by calling her mobile.
> 3. From fs_cli, I execute conference <conf-id> dial Sofia/gateway/xyz/1234567890. Then
> 4. Either she picks up, we her the enter sound and we know she's joined in.
> 5. Or her mobile is switched off, with no voicemail and just an INFO message... And we will never know what happened.
> 
> Is there a way to accomplish the following?
> 
> Have the conference broadcast the early media portion of the call, and still play the enter sound when the called party picks up?
> 
> If this is not possible, I think it would be a great addition to the conferencing module of FS to have a dedicated dial command with support for Early Media.
> 
> Best and thanks,
> Emrah
> --
> Emrah
> 
> ?In theory, theory and practice are the same. In practice, they are not.? Albert Einstein
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lists at kavun.ch  Fri Aug 24 07:40:24 2012
From: lists at kavun.ch (Emrah)
Date: Thu, 23 Aug 2012 23:40:24 -0400
Subject: [Freeswitch-users] Wideband Skype to SIP?
In-Reply-To: <93FD4ECA-6F9A-4C74-8ACE-1FBC8BF405CF@kavun.ch>
References: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>
	<CAKV-jFwM7TKt6HKaKfD4HEG+9KAvDh1J=ZLP+W9dwnVQfb6_Yg@mail.gmail.com>
	<CAKV-jFy2jzH99S4u9G+KL1WkWUQPC1rUEMV7pRLcL7sL-yLUyA@mail.gmail.com>
	<93FD4ECA-6F9A-4C74-8ACE-1FBC8BF405CF@kavun.ch>
Message-ID: <2B2C03B0-B514-42E9-B269-2CEE8752A4D6@kavun.ch>

Got it working on my Squeeze, for some reason I thought it was going to be challenging.

Any idea if we can increase the audio bandwidth? Right now it's giving me an L16 at 16khz, more or less what you get with G722.

Best,
Emrah
On Aug 22, 2012, at 8:07 PM, Emrah <lists at kavun.ch> wrote:

> I think I'll try mod_skypopen. Do you have any how to that lists the appropriate dependencies for Debian?
> 
> Thanks and regards,
> Emrah
> --
> Emrah
> 
> ?In theory, theory and practice are the same. In practice, they are not.? Albert Einstein
> 
> On Aug 22, 2012, at 7:43 PM, Brian Foster <bdfoster at endigotech.com> wrote:
> 
>> P.S. you might want to check this out:
>>  
>> http://www.skype.com/intl/en-us/business/skype-connect#t_pricing
>>  
>> It doesn't mention if you can do wideband with skype users, but you can't do wideband when you connect to the PSTN anyways. Sounds like you might be able to get away with this free if all you want to do is to connect with your skype users. Like I said before, mod_skypopen works well with Debian Squeeze, and we do use it here.
>>  
>> Take it easy,
>>  
>> Brian Foster
>> Endigo Computer LLC
>> 
>> On Wed, Aug 22, 2012 at 7:07 PM, Brian Foster <bdfoster at endigotech.com> wrote:
>> Mod_skypopen works well on Debian Squeeze.
>> 
>> Brian Foster
>> Endigo Computer LLC
>> 
>> Sent from a mobile device.
>> 
>> On Aug 22, 2012 5:05 PM, "Emrah" <lists at kavun.ch> wrote:
>> Hi guys,
>> 
>> I am looking for a SIP based solution to interconnect with Skype, with wideband codec support if possible.
>> 
>> Does that exist?
>> Otherwise, does Skypopen run on Debian Squeeze?
>> 
>> Best,
>> Emrah
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>> 
>> 
>> 
>> -- 
>> Brian D. Foster
>> Endigo Computer LLC
>> Email: bdfoster at endigotech.com
>> Phone: 317-800-7876
>> Indianapolis, Indiana, USA
>> 
>> This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version.
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 gmaruzz at gmail.com  Fri Aug 24 10:30:41 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 24 Aug 2012 08:30:41 +0200
Subject: [Freeswitch-users] Wideband Skype to SIP?
In-Reply-To: <2B2C03B0-B514-42E9-B269-2CEE8752A4D6@kavun.ch>
References: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>
	<CAKV-jFwM7TKt6HKaKfD4HEG+9KAvDh1J=ZLP+W9dwnVQfb6_Yg@mail.gmail.com>
	<CAKV-jFy2jzH99S4u9G+KL1WkWUQPC1rUEMV7pRLcL7sL-yLUyA@mail.gmail.com>
	<93FD4ECA-6F9A-4C74-8ACE-1FBC8BF405CF@kavun.ch>
	<2B2C03B0-B514-42E9-B269-2CEE8752A4D6@kavun.ch>
Message-ID: <CALXCt0oudd=jr6F2bCg5pNpeZOZj0YLfivVTK6DzOXidCTkqyw@mail.gmail.com>

Skype is native at 16 khz SL, you can't have more.

It's much better than pstn or gsm or g711, tough.

Skype imposed hd audio on market, but yes, with FreeSWITCH capable of
48khz... We're spoiled! ;)

-giovanni

On 8/24/12, Emrah <lists at kavun.ch> wrote:
> Got it working on my Squeeze, for some reason I thought it was going to be
> challenging.
>
> Any idea if we can increase the audio bandwidth? Right now it's giving me an
> L16 at 16khz, more or less what you get with G722.
>
> Best,
> Emrah
> On Aug 22, 2012, at 8:07 PM, Emrah <lists at kavun.ch> wrote:
>
>> I think I'll try mod_skypopen. Do you have any how to that lists the
>> appropriate dependencies for Debian?
>>
>> Thanks and regards,
>> Emrah
>> --
>> Emrah
>>
>> ?In theory, theory and practice are the same. In practice, they are not.?
>> Albert Einstein
>>
>> On Aug 22, 2012, at 7:43 PM, Brian Foster <bdfoster at endigotech.com>
>> wrote:
>>
>>> P.S. you might want to check this out:
>>>
>>> http://www.skype.com/intl/en-us/business/skype-connect#t_pricing
>>>
>>> It doesn't mention if you can do wideband with skype users, but you can't
>>> do wideband when you connect to the PSTN anyways. Sounds like you might
>>> be able to get away with this free if all you want to do is to connect
>>> with your skype users. Like I said before, mod_skypopen works well with
>>> Debian Squeeze, and we do use it here.
>>>
>>> Take it easy,
>>>
>>> Brian Foster
>>> Endigo Computer LLC
>>>
>>> On Wed, Aug 22, 2012 at 7:07 PM, Brian Foster <bdfoster at endigotech.com>
>>> wrote:
>>> Mod_skypopen works well on Debian Squeeze.
>>>
>>> Brian Foster
>>> Endigo Computer LLC
>>>
>>> Sent from a mobile device.
>>>
>>> On Aug 22, 2012 5:05 PM, "Emrah" <lists at kavun.ch> wrote:
>>> Hi guys,
>>>
>>> I am looking for a SIP based solution to interconnect with Skype, with
>>> wideband codec support if possible.
>>>
>>> Does that exist?
>>> Otherwise, does Skypopen run on Debian Squeeze?
>>>
>>> Best,
>>> Emrah
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>>
>>> --
>>> Brian D. Foster
>>> Endigo Computer LLC
>>> Email: bdfoster at endigotech.com
>>> Phone: 317-800-7876
>>> Indianapolis, Indiana, USA
>>>
>>> This message contains confidential information and is intended for those
>>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header.
>>> If you are not the intended recipient you are notified that disclosing,
>>> copying, distributing or taking any action in reliance on the contents of
>>> this information is strictly prohibited. E-mail transmission cannot be
>>> guaranteed to be secure or error-free as information could be
>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or
>>> contain viruses. The sender therefore does not accept liability for any
>>> errors or omissions in the contents of this message, which arise as a
>>> result of e-mail transmission. If verification is required please request
>>> a hard-copy version.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jtbock at synacktics.com  Fri Aug 24 07:36:46 2012
From: jtbock at synacktics.com (Tim Bock)
Date: Thu, 23 Aug 2012 21:36:46 -0600
Subject: [Freeswitch-users] freetdm and destination_number
Message-ID: <5036F6CE.3010207@synacktics.com>


Hello,

I'm rather new with FreeSWITCH, migrating over from an asterisk setup.

I have a Digium TDM card, with 3 fxo ports and 1 fxs port.  I have 
everything mostly working, but I'm having an issue with incoming fxo 
calls.  I've found examples where people have used destination_number in 
their condition, but this doesn't seem to work for me.  I can 
successfully use
> <condition field="${channel_name}" expression="^(FreeTDM/2:3/)">

but not

> <condition field="destination_number" expression="(7832889)$"/>
The issue seems to be that destination_number is set to 2 early on, somehow:
> 012-08-23 20:40:43.213780 [DEBUG] ftmod_analog.c:646 [s2c3][1:3] 
> Executing stat
> e handler on 2:3 for RING
> 2012-08-23 20:40:43.213780 [DEBUG] mod_freetdm.c:1992 got FXO sig 2:3 
> [START]
> 2012-08-23 20:40:43.213780 [DEBUG] ftdm_io.c:3131 [s2c3][1:3] Enabled 
> software D
> TMF detector
> 2012-08-23 20:40:43.213780 [DEBUG] mod_freetdm.c:407 Set codec PCMU 20ms
> 2012-08-23 20:40:43.213780 [DEBUG] mod_freetdm.c:1740 Connect inbound 
> channel Fr
> eeTDM/2:3/2
so that later:
> Dialplan: FreeTDM/2:3/2 Regex (FAIL) [fxo_incoming] 
> destination_number(2) =~ /(78328
> 89)$/ break=on-false
Any ideas on what I am doing wrong?  I can make it work using the 
channel_name variable, but using destination_number would be much more 
convenient.

Thank you,
Tim


From vaad.fabi at gmail.com  Fri Aug 24 12:38:47 2012
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Fri, 24 Aug 2012 11:38:47 +0300
Subject: [Freeswitch-users] TLS documentation inaccuracy
In-Reply-To: <CALn233pgL5V1itSiydhX0vV8xVZXzTKgVFD+3+W6dCRAU3x+xQ@mail.gmail.com>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>
	<503226CE.7040307@hw.ac.uk>	<50323D6A.3020708@anew.com.ve>
	<2526639.vT3cT5qGKG@virtex>	<5033C5B1.3070509@hw.ac.uk>
	<5036220C.9080203@gmail.com>
	<CALn233pgL5V1itSiydhX0vV8xVZXzTKgVFD+3+W6dCRAU3x+xQ@mail.gmail.com>
Message-ID: <50373D97.3@gmail.com>

  Thx for information!


On 08/23/2012 07:14 PM, Mitch Capper wrote:
> Vadim update your copy of FS.  The copy you are using is very old so
> the gentls usage has changed 1.2 or git head should give you no
> problem.   A .crt file contains only the certificate the .pem has the
> certificate and private key, just copy the certificate part (its a
> text document) to a file by itself and call .crt to get the .crt.
>
> ~Mitch
>
> On Thu, Aug 23, 2012 at 5:29 AM, vaad.fabi at gmail.com
> <vaad.fabi at gmail.com>  wrote:
>>    Hi all,
>>
>> There are some TLS documentation inaccuracy
>> (http://wiki.freeswitch.org/wiki/Tls):
>>
>> 1.  " ./gentls_cert create_server -cn pbx.freeswitch.org -alt
>> DNS:pbx.freeswitch.org -org freeswitch.org"
>> There is no option "create_server" for gentls_cert , as i can see only
>> setup,create,remove.
>>
>> 2.  "./gentls_cert create_client -cn Client1 -out Client1"
>> There is no option "create_client" for gentls_cert.
>> And there is no exactly information for softphone certificate
>> generation. For Windows we need .crt file certificate, but how it could
>> be generated (.pem only generated) ???
>>
>> Please fix doc and gentls_cert script, or explain all procedures. Thx.
>>
>> --
>> Best Regards,
>> Vadim F.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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


-- 
Best Regards,
Vadim F.



From ntomer at newgen.co.in  Fri Aug 24 12:58:08 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Fri, 24 Aug 2012 14:28:08 +0530
Subject: [Freeswitch-users] Help needed regaridng Contact Center solution
In-Reply-To: <CAGQG5MZFkiSnJhJ6iwuGrYSgtcj5N1JCPatDpVH0rYFg9kSM5w@mail.gmail.com>
References: <004401cd7ba3$f2fbdb10$d8f39130$@co.in>	<1345176410.502dc35a7c3ec@mx.newgen.co.in>	<CAMwZ-t+KUp494GT90dV7oS02sviMVerxJBNCVtyRosg_qneBRA@mail.gmail.com>
	<CAGQG5MZFkiSnJhJ6iwuGrYSgtcj5N1JCPatDpVH0rYFg9kSM5w@mail.gmail.com>
Message-ID: <000001cd81d6$a2fa9cd0$e8efd670$@co.in>

Dear Abdul,

 

Thanks for the pointers. I have installed and configured FreeSWITCH, mod_ivr
and mod_callcenter; and they are working fine.

 

Please tell me how can I take information from caller using the IVR and
insert it in to database. And what is "Using CTI-popup populate client
preferences to the selected agent."?

 

Thanks and Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Abdul
Basit
Sent: Friday, August 17, 2012 1:44 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Help needed regaridng Contact Center
solution

 

You can consider mod_callcenter

http://www.thenoccave.com/2011/10/17/freeswitch-queues-with-mod_callcenter/ 

 

Implement call queues that deal with ACD.

 

Collect information from client using autoattendent system, insert queries
into db and pass call to Queue.

 

Attach few agents to that Queue where any free agent will get this call. 

 

Using CTI-popup populate client preferences to the selected agent.

 

I hope this will help.

 

--

regards,

 

abdul basit  

 

 

On Fri, Aug 17, 2012 at 11:58 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

On Thu, Aug 16, 2012 at 10:06 PM, Nitin Tomer <ntomer at newgen.co.in> wrote:
> Please help.

You're going to need to be more patient than that. You let 17 full
hours pass. You need to give people time to sleep and do their day
jobs.

Anyway, good luck.


Gabe

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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


Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From onyeagbaikenna04 at gmail.com  Fri Aug 24 14:44:00 2012
From: onyeagbaikenna04 at gmail.com (Ikenna Onyeagba)
Date: Fri, 24 Aug 2012 11:44:00 +0100
Subject: [Freeswitch-users] Problem with linking freeswitch with GUI
Message-ID: <CAHibNNek8QZ-J+GzuyxNHhnoJY5XFScW2Kf0q_ES5QmT4YNQBQ@mail.gmail.com>

Hello,
am having problems with downloading freeswitch GUI i have already
installed freeswitch in fedora linux but am stuck from there am using
freeswitch in my final project am implementing voip on it. I want to
link freeswitch with GUI so I can tweak it.
Regards
Ikenna


From lists at kavun.ch  Fri Aug 24 15:19:21 2012
From: lists at kavun.ch (Emrah)
Date: Fri, 24 Aug 2012 07:19:21 -0400
Subject: [Freeswitch-users] Wideband Skype to SIP?
In-Reply-To: <CALXCt0oudd=jr6F2bCg5pNpeZOZj0YLfivVTK6DzOXidCTkqyw@mail.gmail.com>
References: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>
	<CAKV-jFwM7TKt6HKaKfD4HEG+9KAvDh1J=ZLP+W9dwnVQfb6_Yg@mail.gmail.com>
	<CAKV-jFy2jzH99S4u9G+KL1WkWUQPC1rUEMV7pRLcL7sL-yLUyA@mail.gmail.com>
	<93FD4ECA-6F9A-4C74-8ACE-1FBC8BF405CF@kavun.ch>
	<2B2C03B0-B514-42E9-B269-2CEE8752A4D6@kavun.ch>
	<CALXCt0oudd=jr6F2bCg5pNpeZOZj0YLfivVTK6DzOXidCTkqyw@mail.gmail.com>
Message-ID: <A8F6638F-CD26-477E-8BFC-049D99FE4AEB@kavun.ch>

Hi there,

Thanks for your clarifications. FS spoils us indeed!
When I make a computer to computer call, the audio bandwidth is much larger then when the 2 same computers meet in a conf-room via FS @ 32khz.

It sounds like Skype can support higher than L16 @ 16KHZ, how can we confirm? Can FS encode in G7221 at 32000h instead?

Thanks a million,
Emrah
On Aug 24, 2012, at 2:30 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> Skype is native at 16 khz SL, you can't have more.
> 
> It's much better than pstn or gsm or g711, tough.
> 
> Skype imposed hd audio on market, but yes, with FreeSWITCH capable of
> 48khz... We're spoiled! ;)
> 
> -giovanni
> 
> On 8/24/12, Emrah <lists at kavun.ch> wrote:
>> Got it working on my Squeeze, for some reason I thought it was going to be
>> challenging.
>> 
>> Any idea if we can increase the audio bandwidth? Right now it's giving me an
>> L16 at 16khz, more or less what you get with G722.
>> 
>> Best,
>> Emrah
>> On Aug 22, 2012, at 8:07 PM, Emrah <lists at kavun.ch> wrote:
>> 
>>> I think I'll try mod_skypopen. Do you have any how to that lists the
>>> appropriate dependencies for Debian?
>>> 
>>> Thanks and regards,
>>> Emrah
>>> --
>>> Emrah
>>> 
>>> ?In theory, theory and practice are the same. In practice, they are not.?
>>> Albert Einstein
>>> 
>>> On Aug 22, 2012, at 7:43 PM, Brian Foster <bdfoster at endigotech.com>
>>> wrote:
>>> 
>>>> P.S. you might want to check this out:
>>>> 
>>>> http://www.skype.com/intl/en-us/business/skype-connect#t_pricing
>>>> 
>>>> It doesn't mention if you can do wideband with skype users, but you can't
>>>> do wideband when you connect to the PSTN anyways. Sounds like you might
>>>> be able to get away with this free if all you want to do is to connect
>>>> with your skype users. Like I said before, mod_skypopen works well with
>>>> Debian Squeeze, and we do use it here.
>>>> 
>>>> Take it easy,
>>>> 
>>>> Brian Foster
>>>> Endigo Computer LLC
>>>> 
>>>> On Wed, Aug 22, 2012 at 7:07 PM, Brian Foster <bdfoster at endigotech.com>
>>>> wrote:
>>>> Mod_skypopen works well on Debian Squeeze.
>>>> 
>>>> Brian Foster
>>>> Endigo Computer LLC
>>>> 
>>>> Sent from a mobile device.
>>>> 
>>>> On Aug 22, 2012 5:05 PM, "Emrah" <lists at kavun.ch> wrote:
>>>> Hi guys,
>>>> 
>>>> I am looking for a SIP based solution to interconnect with Skype, with
>>>> wideband codec support if possible.
>>>> 
>>>> Does that exist?
>>>> Otherwise, does Skypopen run on Debian Squeeze?
>>>> 
>>>> Best,
>>>> Emrah
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>> 
>>>> 
>>>> 
>>>> --
>>>> Brian D. Foster
>>>> Endigo Computer LLC
>>>> Email: bdfoster at endigotech.com
>>>> Phone: 317-800-7876
>>>> Indianapolis, Indiana, USA
>>>> 
>>>> This message contains confidential information and is intended for those
>>>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header.
>>>> If you are not the intended recipient you are notified that disclosing,
>>>> copying, distributing or taking any action in reliance on the contents of
>>>> this information is strictly prohibited. E-mail transmission cannot be
>>>> guaranteed to be secure or error-free as information could be
>>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or
>>>> contain viruses. The sender therefore does not accept liability for any
>>>> errors or omissions in the contents of this message, which arise as a
>>>> result of e-mail transmission. If verification is required please request
>>>> a hard-copy version.
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lists at kavun.ch  Fri Aug 24 15:45:46 2012
From: lists at kavun.ch (Emrah)
Date: Fri, 24 Aug 2012 07:45:46 -0400
Subject: [Freeswitch-users] Wideband Skype to SIP?
In-Reply-To: <A8F6638F-CD26-477E-8BFC-049D99FE4AEB@kavun.ch>
References: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>
	<CAKV-jFwM7TKt6HKaKfD4HEG+9KAvDh1J=ZLP+W9dwnVQfb6_Yg@mail.gmail.com>
	<CAKV-jFy2jzH99S4u9G+KL1WkWUQPC1rUEMV7pRLcL7sL-yLUyA@mail.gmail.com>
	<93FD4ECA-6F9A-4C74-8ACE-1FBC8BF405CF@kavun.ch>
	<2B2C03B0-B514-42E9-B269-2CEE8752A4D6@kavun.ch>
	<CALXCt0oudd=jr6F2bCg5pNpeZOZj0YLfivVTK6DzOXidCTkqyw@mail.gmail.com>
	<A8F6638F-CD26-477E-8BFC-049D99FE4AEB@kavun.ch>
Message-ID: <943EDB3B-4BBB-4DF1-B592-04D572F34E40@kavun.ch>

The computer to computer call I mention would be using Skype.
On Aug 24, 2012, at 7:19 AM, Emrah <lists at kavun.ch> wrote:

> Hi there,
> 
> Thanks for your clarifications. FS spoils us indeed!
> When I make a computer to computer call, the audio bandwidth is much larger then when the 2 same computers meet in a conf-room via FS @ 32khz.
> 
> It sounds like Skype can support higher than L16 @ 16KHZ, how can we confirm? Can FS encode in G7221 at 32000h instead?
> 
> Thanks a million,
> Emrah
> On Aug 24, 2012, at 2:30 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> 
>> Skype is native at 16 khz SL, you can't have more.
>> 
>> It's much better than pstn or gsm or g711, tough.
>> 
>> Skype imposed hd audio on market, but yes, with FreeSWITCH capable of
>> 48khz... We're spoiled! ;)
>> 
>> -giovanni
>> 
>> On 8/24/12, Emrah <lists at kavun.ch> wrote:
>>> Got it working on my Squeeze, for some reason I thought it was going to be
>>> challenging.
>>> 
>>> Any idea if we can increase the audio bandwidth? Right now it's giving me an
>>> L16 at 16khz, more or less what you get with G722.
>>> 
>>> Best,
>>> Emrah
>>> On Aug 22, 2012, at 8:07 PM, Emrah <lists at kavun.ch> wrote:
>>> 
>>>> I think I'll try mod_skypopen. Do you have any how to that lists the
>>>> appropriate dependencies for Debian?
>>>> 
>>>> Thanks and regards,
>>>> Emrah
>>>> --
>>>> Emrah
>>>> 
>>>> ?In theory, theory and practice are the same. In practice, they are not.?
>>>> Albert Einstein
>>>> 
>>>> On Aug 22, 2012, at 7:43 PM, Brian Foster <bdfoster at endigotech.com>
>>>> wrote:
>>>> 
>>>>> P.S. you might want to check this out:
>>>>> 
>>>>> http://www.skype.com/intl/en-us/business/skype-connect#t_pricing
>>>>> 
>>>>> It doesn't mention if you can do wideband with skype users, but you can't
>>>>> do wideband when you connect to the PSTN anyways. Sounds like you might
>>>>> be able to get away with this free if all you want to do is to connect
>>>>> with your skype users. Like I said before, mod_skypopen works well with
>>>>> Debian Squeeze, and we do use it here.
>>>>> 
>>>>> Take it easy,
>>>>> 
>>>>> Brian Foster
>>>>> Endigo Computer LLC
>>>>> 
>>>>> On Wed, Aug 22, 2012 at 7:07 PM, Brian Foster <bdfoster at endigotech.com>
>>>>> wrote:
>>>>> Mod_skypopen works well on Debian Squeeze.
>>>>> 
>>>>> Brian Foster
>>>>> Endigo Computer LLC
>>>>> 
>>>>> Sent from a mobile device.
>>>>> 
>>>>> On Aug 22, 2012 5:05 PM, "Emrah" <lists at kavun.ch> wrote:
>>>>> Hi guys,
>>>>> 
>>>>> I am looking for a SIP based solution to interconnect with Skype, with
>>>>> wideband codec support if possible.
>>>>> 
>>>>> Does that exist?
>>>>> Otherwise, does Skypopen run on Debian Squeeze?
>>>>> 
>>>>> Best,
>>>>> Emrah
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>> 
>>>>> 
>>>>> 
>>>>> --
>>>>> Brian D. Foster
>>>>> Endigo Computer LLC
>>>>> Email: bdfoster at endigotech.com
>>>>> Phone: 317-800-7876
>>>>> Indianapolis, Indiana, USA
>>>>> 
>>>>> This message contains confidential information and is intended for those
>>>>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header.
>>>>> If you are not the intended recipient you are notified that disclosing,
>>>>> copying, distributing or taking any action in reliance on the contents of
>>>>> this information is strictly prohibited. E-mail transmission cannot be
>>>>> guaranteed to be secure or error-free as information could be
>>>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or
>>>>> contain viruses. The sender therefore does not accept liability for any
>>>>> errors or omissions in the contents of this message, which arise as a
>>>>> result of e-mail transmission. If verification is required please request
>>>>> a hard-copy version.
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 tahir at ictinnovations.com  Fri Aug 24 15:55:11 2012
From: tahir at ictinnovations.com (tahir almas)
Date: Fri, 24 Aug 2012 16:55:11 +0500
Subject: [Freeswitch-users] FreeSWITCH Auto Dialer - Open source
In-Reply-To: <50362C41.8050507@anew.com.ve>
References: <CA+aeUL_D-1sgqxvxshU9oqELh66Ck-_kTxesMCCjnv3DgDu-yg@mail.gmail.com>
	<502D0CAC.8030905@gmail.com>
	<CAJUJwtg7vDr+MDbTvXJJk4HRAr38CTOYY3mRpWwBui0erBXLeg@mail.gmail.com>
	<CAOL0t+LwiTzxXY_VtA6FfN34NzQZYNcBHTHiFfrCK3ZyJMsvYw@mail.gmail.com>
	<CALNZuwnkQ1+nNHECNOM6hs8xCTV1wx3-ucevp01ur8bW5Km1tQ@mail.gmail.com>
	<CA+aeUL9LUZjmJp=LwG2NijHXcZvU3ib+1DTVh1MkPjQapt2Ksw@mail.gmail.com>
	<CAKV-jFxr0b8sJoEto1PBonLGuqQh5GM5GWZkv+5EuO=VRz0AMQ@mail.gmail.com>
	<CALNZuwnv9RwLS2kBwc9x5NMg9TWzvo496FrpXFnPk0nNAxn7Zg@mail.gmail.com>
	<CA+aeUL8kFrwZR=worjfsozAv0Vs+jYLSDGFezvBVQhK3C8Gxcw@mail.gmail.com>
	<50362C41.8050507@anew.com.ve>
Message-ID: <CAOL0t+L8HKe5Euhfvr+K_p-B4PfPgZaoTwFTCD-ZwHuXY=Ozsg@mail.gmail.com>

Hi All

Thanks for pointing out , ICTDialer demo is live and you can browse
application however will not be able to make live test calls

Regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT





On Thu, Aug 23, 2012 at 6:12 PM, Saugort Dario Garcia Tovar <
dgarcia at anew.com.ve> wrote:

>  Hi,
>
> It seems they have fixed the error. I could see the demo
>
>
>
> On 8/23/2012 7:09 AM, Obie Brown wrote:
>
> Yes looks very simple error. When its back I will check demo.
>
>  Thanks everyone for there suggestions.
>
>
>  Thanks,
> Obie Brown
>
>
> On Thu, Aug 23, 2012 at 8:57 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> Brian Foster I'm not worried about it, I'd like to see the demo...
>> -Avi
>>
>>
>> On Thu, Aug 23, 2012 at 2:07 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>>
>>> They might be having issues with their database. Simple error nothing to
>>> worry about.
>>>
>>> Brian Foster
>>> Endigo Computer LLC
>>>
>>> Sent from a mobile device.
>>>  On Aug 23, 2012 4:25 AM, "Obie Brown" <obiebrown at gmail.com> wrote:
>>>
>>>> Yes I was trying to look at demo the there day (few days ago) and there
>>>> was error. Still seems to be error.
>>>>
>>>>  *PDOException*: SQLSTATE[42S02]: Base table or view not found: 1146
>>>> Table 'call4smile.semaphore' doesn't exist: SELECT expire, value FROM
>>>> {semaphore} WHERE name = :name; Array ( [:name] => variable_init ) in *
>>>> lock_may_be_available()* (line *167* of*
>>>> /usr/ictbilling/includes/lock.inc*).
>>>>
>>>>  If you can fix tahir is this something you can fix?
>>>>
>>>>
>>>>  Thanks,
>>>> Obie Brown
>>>>
>>>>
>>>> On Thu, Aug 23, 2012 at 5:45 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>>
>>>>> The demo seems broken: http://demo.ictdialer.org/
>>>>>
>>>>> -Avi
>>>>>
>>>>>
>>>>> On Thu, Aug 23, 2012 at 10:14 AM, tahir almas <
>>>>> tahir at ictinnovations.com> wrote:
>>>>>
>>>>>> I will recommend to check ICTDialer http://www.ictdialer.org that is
>>>>>> based on Plivo Communication framework / Freeswitch and support for
>>>>>> standard voice broadcasting as well as Interactive Voice broadcasting also
>>>>>> you have support IVR Designer to design your IVR with drag and drop feature
>>>>>>
>>>>>> Regards
>>>>>>
>>>>>> *Tahir Almas*
>>>>>>
>>>>>> Managing Partner
>>>>>> ICT Innovations
>>>>>> http://www.ictinnovations.com
>>>>>> Leveraging open source in ICT
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Aug 16, 2012 at 11:41 PM, SamyGo <govoiper at gmail.com> wrote:
>>>>>>
>>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> Well I'd say to figure out your requirements first. I once got into
>>>>>>> same situation as your's and believe me reinventing the wheel (tailor made
>>>>>>> solution) for exactly the requirement was the best thing. Atleast I know
>>>>>>> I've created something which is doing all that I needed and the time spent
>>>>>>> on this was less than a week !!
>>>>>>>
>>>>>>> Its upto you and your requirements to choose the solution.
>>>>>>>
>>>>>>> BR
>>>>>>> Sammy Go.
>>>>>>>  On Aug 16, 2012 8:11 PM, "Abaci" <abaci64 at gmail.com> wrote:
>>>>>>>
>>>>>>>>  http://www.ictdialer.org/
>>>>>>>>
>>>>>>>> On 8/16/2012 9:53 AM, Obie Brown wrote:
>>>>>>>>
>>>>>>>> Hi,
>>>>>>>>
>>>>>>>>  I really want to work with an Open Source dialer and FreeSWITCH
>>>>>>>> but the only thing I can find for this is Newfies Dialer which does not
>>>>>>>> have features of a Auto Dialer as I need. Also its use for Voice
>>>>>>>> Broadcasting is limited.
>>>>>>>>
>>>>>>>>  My question is. What other Auto Dialers that run on FreeSWITCH
>>>>>>>> are there. If any.
>>>>>>>>
>>>>>>>>
>>>>>>>>  Thanks,
>>>>>>>> Obie Brown
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>>>>>>
>>>>>>>> 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
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.2197 / Virus Database: 2437/5219 - Release Date: 08/23/12
>
>
>
> --
> Atentamente,
> *Dario Garc?a*
> Consultor.
>
> CCCT, Nivel C2, Sector Yarey, Mz,
> Ofc. MZ03a.
> Caracas-Venezuela.
> Tel?fono: +58 212 9081842
> Cel: +58 412 2221515
> dgarcia at anew.com.ve
> http://www.anew.com.ve
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveu at coppice.org  Fri Aug 24 15:59:20 2012
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 24 Aug 2012 19:59:20 +0800
Subject: [Freeswitch-users] Wideband Skype to SIP?
In-Reply-To: <CALXCt0oudd=jr6F2bCg5pNpeZOZj0YLfivVTK6DzOXidCTkqyw@mail.gmail.com>
References: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>
	<CAKV-jFwM7TKt6HKaKfD4HEG+9KAvDh1J=ZLP+W9dwnVQfb6_Yg@mail.gmail.com>
	<CAKV-jFy2jzH99S4u9G+KL1WkWUQPC1rUEMV7pRLcL7sL-yLUyA@mail.gmail.com>
	<93FD4ECA-6F9A-4C74-8ACE-1FBC8BF405CF@kavun.ch>
	<2B2C03B0-B514-42E9-B269-2CEE8752A4D6@kavun.ch>
	<CALXCt0oudd=jr6F2bCg5pNpeZOZj0YLfivVTK6DzOXidCTkqyw@mail.gmail.com>
Message-ID: <50376C98.8070906@coppice.org>

Hi Giovanni,

On 08/24/2012 02:30 PM, Giovanni Maruzzelli wrote:
> Skype is native at 16 khz SL, you can't have more.
Is that a limitation of Skype on Linux, or something? The SILK codec 
uses a sample rate of 24ksps, so if everything in the Skype world were 
limited to 16ksps they wouldn't be getting the full benefit of their key 
codec.

Steve



From lists at kavun.ch  Fri Aug 24 16:01:31 2012
From: lists at kavun.ch (Emrah)
Date: Fri, 24 Aug 2012 08:01:31 -0400
Subject: [Freeswitch-users] Wideband Skype to SIP?
In-Reply-To: <943EDB3B-4BBB-4DF1-B592-04D572F34E40@kavun.ch>
References: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>
	<CAKV-jFwM7TKt6HKaKfD4HEG+9KAvDh1J=ZLP+W9dwnVQfb6_Yg@mail.gmail.com>
	<CAKV-jFy2jzH99S4u9G+KL1WkWUQPC1rUEMV7pRLcL7sL-yLUyA@mail.gmail.com>
	<93FD4ECA-6F9A-4C74-8ACE-1FBC8BF405CF@kavun.ch>
	<2B2C03B0-B514-42E9-B269-2CEE8752A4D6@kavun.ch>
	<CALXCt0oudd=jr6F2bCg5pNpeZOZj0YLfivVTK6DzOXidCTkqyw@mail.gmail.com>
	<A8F6638F-CD26-477E-8BFC-049D99FE4AEB@kavun.ch>
	<943EDB3B-4BBB-4DF1-B592-04D572F34E40@kavun.ch>
Message-ID: <8BE85234-2A78-4347-9164-B2EE91A22846@kavun.ch>

I read the SVOPC Wikipedia page and it does say it's sampled at 16khz indeed.
I do encourage you to try to make a regular Skype call to echo123 and then try the same routed through FS. The quality is substantially different and I do think the audio bandwidth remains higher on a Skype to Skype call.
Since the support for FS is done through an emulation driver, it would be good to try different encoding settings to confirm.

It might also be the version of Skype that is running that could be limited at L16 / 16khz. The newer version do support Sylk.

Best,
Emrah
On Aug 24, 2012, at 7:45 AM, Emrah <lists at kavun.ch> wrote:

> The computer to computer call I mention would be using Skype.
> On Aug 24, 2012, at 7:19 AM, Emrah <lists at kavun.ch> wrote:
> 
>> Hi there,
>> 
>> Thanks for your clarifications. FS spoils us indeed!
>> When I make a computer to computer call, the audio bandwidth is much larger then when the 2 same computers meet in a conf-room via FS @ 32khz.
>> 
>> It sounds like Skype can support higher than L16 @ 16KHZ, how can we confirm? Can FS encode in G7221 at 32000h instead?
>> 
>> Thanks a million,
>> Emrah
>> On Aug 24, 2012, at 2:30 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>> 
>>> Skype is native at 16 khz SL, you can't have more.
>>> 
>>> It's much better than pstn or gsm or g711, tough.
>>> 
>>> Skype imposed hd audio on market, but yes, with FreeSWITCH capable of
>>> 48khz... We're spoiled! ;)
>>> 
>>> -giovanni
>>> 
>>> On 8/24/12, Emrah <lists at kavun.ch> wrote:
>>>> Got it working on my Squeeze, for some reason I thought it was going to be
>>>> challenging.
>>>> 
>>>> Any idea if we can increase the audio bandwidth? Right now it's giving me an
>>>> L16 at 16khz, more or less what you get with G722.
>>>> 
>>>> Best,
>>>> Emrah
>>>> On Aug 22, 2012, at 8:07 PM, Emrah <lists at kavun.ch> wrote:
>>>> 
>>>>> I think I'll try mod_skypopen. Do you have any how to that lists the
>>>>> appropriate dependencies for Debian?
>>>>> 
>>>>> Thanks and regards,
>>>>> Emrah
>>>>> --
>>>>> Emrah
>>>>> 
>>>>> ?In theory, theory and practice are the same. In practice, they are not.?
>>>>> Albert Einstein
>>>>> 
>>>>> On Aug 22, 2012, at 7:43 PM, Brian Foster <bdfoster at endigotech.com>
>>>>> wrote:
>>>>> 
>>>>>> P.S. you might want to check this out:
>>>>>> 
>>>>>> http://www.skype.com/intl/en-us/business/skype-connect#t_pricing
>>>>>> 
>>>>>> It doesn't mention if you can do wideband with skype users, but you can't
>>>>>> do wideband when you connect to the PSTN anyways. Sounds like you might
>>>>>> be able to get away with this free if all you want to do is to connect
>>>>>> with your skype users. Like I said before, mod_skypopen works well with
>>>>>> Debian Squeeze, and we do use it here.
>>>>>> 
>>>>>> Take it easy,
>>>>>> 
>>>>>> Brian Foster
>>>>>> Endigo Computer LLC
>>>>>> 
>>>>>> On Wed, Aug 22, 2012 at 7:07 PM, Brian Foster <bdfoster at endigotech.com>
>>>>>> wrote:
>>>>>> Mod_skypopen works well on Debian Squeeze.
>>>>>> 
>>>>>> Brian Foster
>>>>>> Endigo Computer LLC
>>>>>> 
>>>>>> Sent from a mobile device.
>>>>>> 
>>>>>> On Aug 22, 2012 5:05 PM, "Emrah" <lists at kavun.ch> wrote:
>>>>>> Hi guys,
>>>>>> 
>>>>>> I am looking for a SIP based solution to interconnect with Skype, with
>>>>>> wideband codec support if possible.
>>>>>> 
>>>>>> Does that exist?
>>>>>> Otherwise, does Skypopen run on Debian Squeeze?
>>>>>> 
>>>>>> Best,
>>>>>> Emrah
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> --
>>>>>> Brian D. Foster
>>>>>> Endigo Computer LLC
>>>>>> Email: bdfoster at endigotech.com
>>>>>> Phone: 317-800-7876
>>>>>> Indianapolis, Indiana, USA
>>>>>> 
>>>>>> This message contains confidential information and is intended for those
>>>>>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header.
>>>>>> If you are not the intended recipient you are notified that disclosing,
>>>>>> copying, distributing or taking any action in reliance on the contents of
>>>>>> this information is strictly prohibited. E-mail transmission cannot be
>>>>>> guaranteed to be secure or error-free as information could be
>>>>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or
>>>>>> contain viruses. The sender therefore does not accept liability for any
>>>>>> errors or omissions in the contents of this message, which arise as a
>>>>>> result of e-mail transmission. If verification is required please request
>>>>>> a hard-copy version.
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>> 
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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 onyeagbaikenna04 at gmail.com  Fri Aug 24 16:12:04 2012
From: onyeagbaikenna04 at gmail.com (onyeagbaikenna04 at gmail.com)
Date: Fri, 24 Aug 2012 12:12:04 +0000
Subject: [Freeswitch-users] Wideband Skype to SIP?
In-Reply-To: <A8F6638F-CD26-477E-8BFC-049D99FE4AEB@kavun.ch>
References: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>
	<CAKV-jFwM7TKt6HKaKfD4HEG+9KAvDh1J=ZLP+W9dwnVQfb6_Yg@mail.gmail.com>
	<CAKV-jFy2jzH99S4u9G+KL1WkWUQPC1rUEMV7pRLcL7sL-yLUyA@mail.gmail.com>
	<93FD4ECA-6F9A-4C74-8ACE-1FBC8BF405CF@kavun.ch>
	<2B2C03B0-B514-42E9-B269-2CEE8752A4D6@kavun.ch>
	<CALXCt0oudd=jr6F2bCg5pNpeZOZj0YLfivVTK6DzOXidCTkqyw@mail.gmail.com>
	<A8F6638F-CD26-477E-8BFC-049D99FE4AEB@kavun.ch>
Message-ID: <677381029-1345810230-cardhu_decombobulator_blackberry.rim.net-2095477982-@b13.c16.bise7.blackberry>

Hi.
I have installed freeswitch on linux, pls can u tell me how to connect a GUI to d installed freeswitch. Am implementing voip on freeswitch. Thanks
Sent from my BlackBerry smartphone from Virgin Media

-----Original Message-----
From: Emrah <lists at kavun.ch>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Fri, 24 Aug 2012 07:19:21 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Wideband Skype to SIP?

Hi there,

Thanks for your clarifications. FS spoils us indeed!
When I make a computer to computer call, the audio bandwidth is much larger then when the 2 same computers meet in a conf-room via FS @ 32khz.

It sounds like Skype can support higher than L16 @ 16KHZ, how can we confirm? Can FS encode in G7221 at 32000h instead?

Thanks a million,
Emrah
On Aug 24, 2012, at 2:30 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> Skype is native at 16 khz SL, you can't have more.
> 
> It's much better than pstn or gsm or g711, tough.
> 
> Skype imposed hd audio on market, but yes, with FreeSWITCH capable of
> 48khz... We're spoiled! ;)
> 
> -giovanni
> 
> On 8/24/12, Emrah <lists at kavun.ch> wrote:
>> Got it working on my Squeeze, for some reason I thought it was going to be
>> challenging.
>> 
>> Any idea if we can increase the audio bandwidth? Right now it's giving me an
>> L16 at 16khz, more or less what you get with G722.
>> 
>> Best,
>> Emrah
>> On Aug 22, 2012, at 8:07 PM, Emrah <lists at kavun.ch> wrote:
>> 
>>> I think I'll try mod_skypopen. Do you have any how to that lists the
>>> appropriate dependencies for Debian?
>>> 
>>> Thanks and regards,
>>> Emrah
>>> --
>>> Emrah
>>> 
>>> ?In theory, theory and practice are the same. In practice, they are not.?
>>> Albert Einstein
>>> 
>>> On Aug 22, 2012, at 7:43 PM, Brian Foster <bdfoster at endigotech.com>
>>> wrote:
>>> 
>>>> P.S. you might want to check this out:
>>>> 
>>>> http://www.skype.com/intl/en-us/business/skype-connect#t_pricing
>>>> 
>>>> It doesn't mention if you can do wideband with skype users, but you can't
>>>> do wideband when you connect to the PSTN anyways. Sounds like you might
>>>> be able to get away with this free if all you want to do is to connect
>>>> with your skype users. Like I said before, mod_skypopen works well with
>>>> Debian Squeeze, and we do use it here.
>>>> 
>>>> Take it easy,
>>>> 
>>>> Brian Foster
>>>> Endigo Computer LLC
>>>> 
>>>> On Wed, Aug 22, 2012 at 7:07 PM, Brian Foster <bdfoster at endigotech.com>
>>>> wrote:
>>>> Mod_skypopen works well on Debian Squeeze.
>>>> 
>>>> Brian Foster
>>>> Endigo Computer LLC
>>>> 
>>>> Sent from a mobile device.
>>>> 
>>>> On Aug 22, 2012 5:05 PM, "Emrah" <lists at kavun.ch> wrote:
>>>> Hi guys,
>>>> 
>>>> I am looking for a SIP based solution to interconnect with Skype, with
>>>> wideband codec support if possible.
>>>> 
>>>> Does that exist?
>>>> Otherwise, does Skypopen run on Debian Squeeze?
>>>> 
>>>> Best,
>>>> Emrah
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>> 
>>>> 
>>>> 
>>>> --
>>>> Brian D. Foster
>>>> Endigo Computer LLC
>>>> Email: bdfoster at endigotech.com
>>>> Phone: 317-800-7876
>>>> Indianapolis, Indiana, USA
>>>> 
>>>> This message contains confidential information and is intended for those
>>>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header.
>>>> If you are not the intended recipient you are notified that disclosing,
>>>> copying, distributing or taking any action in reliance on the contents of
>>>> this information is strictly prohibited. E-mail transmission cannot be
>>>> guaranteed to be secure or error-free as information could be
>>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or
>>>> contain viruses. The sender therefore does not accept liability for any
>>>> errors or omissions in the contents of this message, which arise as a
>>>> result of e-mail transmission. If verification is required please request
>>>> a hard-copy version.
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 tahir at ictinnovations.com  Fri Aug 24 16:32:23 2012
From: tahir at ictinnovations.com (tahir almas)
Date: Fri, 24 Aug 2012 17:32:23 +0500
Subject: [Freeswitch-users] Voice broadcasting
In-Reply-To: <CALNZuwkobxi_6pVf14HH6P7MfMTD7eFLuP4tYmrPQvH6R7JiDQ@mail.gmail.com>
References: <CAAdTMa-Qz1UyH+wXKEGXRBHF=Hjh7S_jpxe_BSphXgsrYpo_aw@mail.gmail.com>
	<CC359325.3A8AE%krice@freeswitch.org>
	<CAKV-jFzQizHQAbUEQxCPdi_Wc9odhcL0j7oc9grHX=kceZ3-aQ@mail.gmail.com>
	<CAAdTMa-wwTiyZ+=caecU8mC5K8Z2A_2Atnq4yonDRz-172BU5g@mail.gmail.com>
	<CAOL0t+LYhOBid83cWPwvUEPqyE309Vm6DC7ZkoQWO42CVUp4Og@mail.gmail.com>
	<CAKV-jFykvFmKSPf1ZGL0tdth2ZJPyk2tX0J2aiWJHYW5Zg7vRA@mail.gmail.com>
	<CAPoPLj_QVqmA43PWM9rpzWrAii3k_3kp4BhYxngip+A84NbQAw@mail.gmail.com>
	<CAKzWOxU8qtor5ZkENWdncGp89YP7vFHSQJCxD03y3zK-+chM0w@mail.gmail.com>
	<CALNZuwkobxi_6pVf14HH6P7MfMTD7eFLuP4tYmrPQvH6R7JiDQ@mail.gmail.com>
Message-ID: <CAOL0t+LZ2fgh9CKpM+ncg-C9c_rHK=sYinPYRq15UF7AiYShgw@mail.gmail.com>

Last day I checked the list and posted these two comments where appropriate

The interesting thing I like to mention that  we have commercial product
competitive to ICTDialer,  entirely  different in design  as compared to
ICTDialer  and we use commercial channels to market it rather than this
mailing list

ICTDialer is purely OSS contribution for community  based on Open Source
communication framework Plivo / freeswitch and MySQL as back-end and
re-known Open Source CMS  Drupal as Front-End  with no hidden design  and
people here still have objection on posting ?

Regards

*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT





On Thu, Aug 23, 2012 at 8:57 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Since ICT Dialer was mentioned only on two threads where it was related,
> and it is indeed OSS, it seems entirely appropriate.
>
> Their download page http://www.ictdialer.org/node/7 ->
> https://github.com/ictinnovations/ictdialer
>
> -Avi
>
>
> On Thu, Aug 23, 2012 at 6:51 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Thu, Aug 23, 2012 at 5:37 AM, David J <david at styleflare.com> wrote:
>>
>>> Brian.
>>>
>>> I will agree with you only if ICT dialer is a commercial product.
>>>
>>> Otherwise to mention it on list should be OK as long as its free switch
>>> related.
>>>
>> We are pretty tolerant of people mentioning their FreeSWITCH-based
>> commercial products. However, we don't want the list to turn into a
>> marketing free-for-all. If you have a commercial product that you think
>> would meet the OP's needs then it's okay to mention it. The caveat is that
>> you really must also list any [FL]OSS options as well, even if those free
>> options are less complete than your own commercial solution.
>>
>> Here's an example:
>>
>> Q: "What's the best GUI for FreeSWITCH?"
>>
>> A: "IMHO, the best GUI is the CudaTel. More info available at
>> www.cudatel.com. If you are looking for an OSS alternative you can check
>> out Blue.Box (2600hz.org) or FusionPBX (fusionpbx.com) and try those
>> out. I've also heard of FreePyBX (freepybx.org) but I know very little
>> about that one."
>>
>> If you keep it balanced, reasonable, and non-spammy then you'll be fine.
>> :)
>>
>> -MC
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 dgarcia at anew.com.ve  Fri Aug 24 18:29:57 2012
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Fri, 24 Aug 2012 09:59:57 -0430
Subject: [Freeswitch-users] Compilation error on debian
Message-ID: <50378FE5.2010608@anew.com.ve>

Hi guys,


I have got a error trying to compile FS in debian.

I have configured a small virtual machine, installed debian 6.2. I pull 
FS from git. I did bootstrap and compile. When I invoke make I got this:

make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" 
/usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq 
)"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all 
; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then 
tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e 
"s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in 
$tmp_mods ; do echo $i-clean ; done )"; echo $mods )" 
"OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" 
/usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq 
)"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo 
$i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z 
"" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf 
| sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; 
mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods 
)" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" 
/usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e 
"s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all 
; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep 
"#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e 
"s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo 
$i-clean ; done )"; echo $mods )" 
"OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" 
/usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e 
"s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo 
$i-install ; done)"; echo $mods )" 
"OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" 
/usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e 
"s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo 
$i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
quiet_libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I./src/include 
-I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include 
-I/usr/local/src/freeswitch/src/include 
-I/usr/local/src/freeswitch/src/include 
-I/usr/local/src/freeswitch/libs/libteletone/src -fPIC 
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 
-pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE 
-I/usr/local/src/freeswitch/libs/apr/include 
-I/usr/local/src/freeswitch/libs/apr-util/include 
-I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib 
-I/usr/local/src/freeswitch/libs/stfu 
-I/usr/local/src/freeswitch/libs/sqlite 
-I/usr/local/src/freeswitch/libs/pcre 
-I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include 
-I/usr/local/src/freeswitch/libs/srtp/include 
-I/usr/local/src/freeswitch/libs/srtp/crypto/include 
-Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src 
-I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -DENABLE_SRTP 
-DSWITCH_HAVE_ODBC -I/usr/include 
-I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT 
-I/usr/local/src/freeswitch/libs/curl/include 
-I/usr/local/src/freeswitch/src/include 
-I/usr/local/src/freeswitch/src/include 
-I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror 
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g 
-ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic 
-Wdeclaration-after-statement -g -O2 -MT 
libfreeswitch_la-switch_core_session.lo -MD -MP -MF 
.deps/libfreeswitch_la-switch_core_session.Tpo -c 
src/switch_core_session.c  -fPIC -DPIC -o 
.libs/libfreeswitch_la-switch_core_session.o
*cc1: warnings being treated as errors
src/switch_core_session.c: In function 
'switch_core_session_thread_pool_worker':
src/switch_core_session.c:1478: error: format '%ld' expects type 'long 
int', but argument 9 has type 'switch_size_t'
src/switch_core_session.c:1483: error: format '%ld' expects type 'long 
int', but argument 9 has type 'switch_size_t'
make[1]: *** [libfreeswitch_la-switch_core_session.lo] Error 1
make: *** [all] Error 2
*
What is wrong?

Thanks in advance

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From jeff at jefflenk.com  Fri Aug 24 18:38:55 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 24 Aug 2012 07:38:55 -0700 (PDT)
Subject: [Freeswitch-users] Compilation error on debian
In-Reply-To: <50378FE5.2010608@anew.com.ve>
References: <50378FE5.2010608@anew.com.ve>
Message-ID: <1345819135651-7582171.post@n2.nabble.com>

update again



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compilation-error-on-debian-tp7582170p7582171.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From eranga.erl at gmail.com  Fri Aug 24 14:51:55 2012
From: eranga.erl at gmail.com (Eranga Udesh)
Date: Fri, 24 Aug 2012 16:21:55 +0530
Subject: [Freeswitch-users] Mod Erlang Event in Freeswitch
In-Reply-To: <CALg+e6z0=xhmohAJHEoapsP4oGgBzg1uyQgBe0NDJSeRrEa9RA@mail.gmail.com>
References: <CALg+e6z0=xhmohAJHEoapsP4oGgBzg1uyQgBe0NDJSeRrEa9RA@mail.gmail.com>
Message-ID: <CALg+e6zRTn-=znDxQMhsTEojXCg+T8MhYcPHFbYSrRJirvq_vw@mail.gmail.com>

To answer my own question, the issue was with the Erlang version. It seels
Mod Erlang Event is not compatible with R15B01 version. Once I downgrade
the module compiling Erlang version to R14B03, it works fine.

Hope it helps others in the list.

- Eranga



On Thu, Aug 23, 2012 at 5:32 PM, Eranga Udesh <eranga.erl at gmail.com> wrote:

> Hi,
>
> I'm trying mod_erlang_event module in Freeswitch. However when a call
> comes to extension 111, the Erlang node receives RPC call correctly (my
> config as below) but it seems Freeswitch module does not receive the
> returned {Ref, NewPid} tuple of the Erlang function. I use Erlang R15B01
> version.
>
> I also tried with Registered process configuration as in the Mod Erlang
> Event wiki, but there also it timeout. I wonder if it is an incompatibility
> with Erlang 15B?
>
> Any ideas what could be wrong?
>
> Config:
>
> <extension name="to_erlang" continue="true">
>    <condition field="destination_number" expression="^111$" >
>        <action application="erlang" data="fsivr:new_call ivr at linux" />
>    </condition>
> </extension>
>
> Output in freeswitch debug:
>
> 2012-08-23 17:06:34.057997 [DEBUG] mod_erlang_event.c:1519 enter
> erlang_outbound_function fsivr:new_call ivr at linux
> 2012-08-23 17:06:34.057997 [DEBUG] mod_erlang_event.c:1525 Creating new
> listener for session
> 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1527 Launching new
> listener
> 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1537 Creating new
> spawned session for listener
> 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1423 rpc call:
> fsivr:new_call(Ref)
> 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1008 Connection Open
> 2012-08-23 17:06:39.177997 [WARNING] mod_erlang_event.c:1436 Timed out
> when waiting for outbound pid 12.0.0 at freeswitch@linux
> 404fc3c7-45d1-4194-ad37-1e118fdee8d0
> 2012-08-23 17:06:39.177997 [DEBUG] switch_channel.c:2919
> (sofia/internal/5000 at linux4) Callstate Change RINGING -> HANGUP
>
> Thanks in advance.
> - Eranga
>
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From mike at jerris.com  Fri Aug 24 18:53:06 2012
From: mike at jerris.com (Michael Jerris)
Date: Fri, 24 Aug 2012 10:53:06 -0400
Subject: [Freeswitch-users] Compilation error on debian
In-Reply-To: <50378FE5.2010608@anew.com.ve>
References: <50378FE5.2010608@anew.com.ve>
Message-ID: <75A08564-4DBC-410F-B679-6CF8E94FA7B9@jerris.com>

This is a bug, could you please file it at http://jira.freeswitch.org

Thanks
Mike

On Aug 24, 2012, at 10:29 AM, Saugort Dario Garcia Tovar <dgarcia at anew.com.ve> wrote:

> Hi guys,
> 
> 
> I have got a error trying to compile FS in debian.
> 
> I have configured a small virtual machine, installed debian 6.2. I pull FS from git. I did bootstrap and compile. When I invoke make I got this:
> 
> make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z     "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )";  mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive
> quiet_libtool: compile:  gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_core_session.lo -MD -MP -MF .deps/libfreeswitch_la-switch_core_session.Tpo -c src/switch_core_session.c  -fPIC -DPIC -o .libs/libfreeswitch_la-switch_core_session.o
> cc1: warnings being treated as errors
> src/switch_core_session.c: In function ?switch_core_session_thread_pool_worker?:
> src/switch_core_session.c:1478: error: format ?%ld? expects type ?long int?, but argument 9 has type ?switch_size_t?
> src/switch_core_session.c:1483: error: format ?%ld? expects type ?long int?, but argument 9 has type ?switch_size_t?
> make[1]: *** [libfreeswitch_la-switch_core_session.lo] Error 1
> make: *** [all] Error 2

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From msc at freeswitch.org  Fri Aug 24 19:39:40 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 24 Aug 2012 08:39:40 -0700
Subject: [Freeswitch-users] Voice broadcasting
In-Reply-To: <CAOL0t+LZ2fgh9CKpM+ncg-C9c_rHK=sYinPYRq15UF7AiYShgw@mail.gmail.com>
References: <CAAdTMa-Qz1UyH+wXKEGXRBHF=Hjh7S_jpxe_BSphXgsrYpo_aw@mail.gmail.com>
	<CC359325.3A8AE%krice@freeswitch.org>
	<CAKV-jFzQizHQAbUEQxCPdi_Wc9odhcL0j7oc9grHX=kceZ3-aQ@mail.gmail.com>
	<CAAdTMa-wwTiyZ+=caecU8mC5K8Z2A_2Atnq4yonDRz-172BU5g@mail.gmail.com>
	<CAOL0t+LYhOBid83cWPwvUEPqyE309Vm6DC7ZkoQWO42CVUp4Og@mail.gmail.com>
	<CAKV-jFykvFmKSPf1ZGL0tdth2ZJPyk2tX0J2aiWJHYW5Zg7vRA@mail.gmail.com>
	<CAPoPLj_QVqmA43PWM9rpzWrAii3k_3kp4BhYxngip+A84NbQAw@mail.gmail.com>
	<CAKzWOxU8qtor5ZkENWdncGp89YP7vFHSQJCxD03y3zK-+chM0w@mail.gmail.com>
	<CALNZuwkobxi_6pVf14HH6P7MfMTD7eFLuP4tYmrPQvH6R7JiDQ@mail.gmail.com>
	<CAOL0t+LZ2fgh9CKpM+ncg-C9c_rHK=sYinPYRq15UF7AiYShgw@mail.gmail.com>
Message-ID: <CAKzWOxVrFetN6_+pQ5U4G0vqxT-VQWpumK+tt5D5vBZK-n2Wpg@mail.gmail.com>

Tahir,

Thanks for the clarification. This is totally cool. We <3 OSS! :)
-MC

On Fri, Aug 24, 2012 at 5:32 AM, tahir almas <tahir at ictinnovations.com>wrote:

> Last day I checked the list and posted these two comments where
> appropriate
>
> The interesting thing I like to mention that  we have commercial product
> competitive to ICTDialer,  entirely  different in design  as compared to
> ICTDialer  and we use commercial channels to market it rather than this
> mailing list
>
> ICTDialer is purely OSS contribution for community  based on Open Source
> communication framework Plivo / freeswitch and MySQL as back-end and
> re-known Open Source CMS  Drupal as Front-End  with no hidden design  and
> people here still have objection on posting ?
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Fri Aug 24 19:41:00 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 24 Aug 2012 08:41:00 -0700
Subject: [Freeswitch-users] Problem with linking freeswitch with GUI
In-Reply-To: <CAHibNNek8QZ-J+GzuyxNHhnoJY5XFScW2Kf0q_ES5QmT4YNQBQ@mail.gmail.com>
References: <CAHibNNek8QZ-J+GzuyxNHhnoJY5XFScW2Kf0q_ES5QmT4YNQBQ@mail.gmail.com>
Message-ID: <CAKzWOxVM2XGehpXYzBzjTGEzicsGovoj7k_A1gVuDttCng=5YQ@mail.gmail.com>

Which GUI are you trying to use?
-MC

On Fri, Aug 24, 2012 at 3:44 AM, Ikenna Onyeagba <onyeagbaikenna04 at gmail.com
> wrote:

> Hello,
> am having problems with downloading freeswitch GUI i have already
> installed freeswitch in fedora linux but am stuck from there am using
> freeswitch in my final project am implementing voip on it. I want to
> link freeswitch with GUI so I can tweak it.
> Regards
> Ikenna
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From urealfrank at gmail.com  Fri Aug 24 19:42:11 2012
From: urealfrank at gmail.com (urealfrank)
Date: Fri, 24 Aug 2012 16:42:11 +0100
Subject: [Freeswitch-users] FS 1.2.0 fails to build on OS X 10.8
Message-ID: <1C56216DA9394B4084D79C2B3E4F2F01@gmail.com>


Hi,

Anyone had success building FS 1.2.0 on Mountain Lion?

I did configure it this way:

$ git pull ; make clean ; ./bootstrap.sh ; ./configure --prefix=/usr/local/fs120 --disable-dependency-tracking --with-odbc=/usr/local/Cellar/unixodbc/2.3.1 --with-openssl ; make

?
Making all in ipt
Making all in sdp
Making all in url
Making all in msg
Making all in sip
Making all in http
Making all in soa
Making all in tport
LTCOMPILE tport_tls.lo
cc1: warnings being treated as errors
tport_tls.c: In function 'tls_init_once':
tport_tls.c:98: warning: 'SSL_library_init' is deprecated (declared at /usr/include/openssl/ssl.h:1553)
tport_tls.c:99: warning: 'SSL_load_error_strings' is deprecated (declared at /usr/include/openssl/ssl.h:1416)
tport_tls.c:100: warning: 'SSL_get_ex_new_index' is deprecated (declared at /usr/include/openssl/ssl.h:1589)
tport_tls.c: In function 'tls_log_errors':
tport_tls.c:144: warning: 'ERR_get_error' is deprecated (declared at /usr/include/openssl/err.h:266)
tport_tls.c:151: warning: 'ERR_get_error' is deprecated (declared at /usr/include/openssl/err.h:266)
tport_tls.c:153: warning: 'ERR_lib_error_string' is deprecated (declared at /usr/include/openssl/err.h:281)
tport_tls.c:154: warning: 'ERR_func_error_string' is deprecated (declared at /usr/include/openssl/err.h:282)
tport_tls.c:155: warning: 'ERR_reason_error_string' is deprecated (declared at /usr/include/openssl/err.h:283)
tport_tls.c: In function 'tls_verify_cb':
tport_tls.c:220: warning: 'X509_STORE_CTX_get_current_cert' is deprecated (declared at /usr/include/openssl/x509_vfy.h:454)
tport_tls.c:221: warning: 'X509_STORE_CTX_get_error_depth' is deprecated (declared at /usr/include/openssl/x509_vfy.h:453)
tport_tls.c:222: warning: 'X509_STORE_CTX_get_error' is deprecated (declared at /usr/include/openssl/x509_vfy.h:451)
tport_tls.c:223: warning: 'SSL_get_ex_data_X509_STORE_CTX_idx' is deprecated (declared at /usr/include/openssl/ssl.h:1601)
tport_tls.c:224: warning: 'X509_STORE_CTX_get_ex_data' is deprecated (declared at /usr/include/openssl/x509_vfy.h:450)
tport_tls.c:225: warning: 'SSL_get_ex_data' is deprecated (declared at /usr/include/openssl/ssl.h:1587)
tport_tls.c:237: warning: 'X509_STORE_CTX_set_error' is deprecated (declared at /usr/include/openssl/x509_vfy.h:452)
tport_tls.c:239: warning: 'X509_STORE_CTX_set_error' is deprecated (declared at /usr/include/openssl/x509_vfy.h:452)
tport_tls.c:246: warning: 'X509_STORE_CTX_set_error' is deprecated (declared at /usr/include/openssl/x509_vfy.h:452)
tport_tls.c:254: warning: 'X509_NAME_oneline' is deprecated (declared at /usr/include/openssl/x509.h:984)
tport_tls.c:254: warning: 'X509_get_issuer_name' is deprecated (declared at /usr/include/openssl/x509.h:1011)
tport_tls.c:256: warning: 'X509_NAME_oneline' is deprecated (declared at /usr/include/openssl/x509.h:984)
tport_tls.c:256: warning: 'X509_get_subject_name' is deprecated (declared at /usr/include/openssl/x509.h:1013)
tport_tls.c:258: warning: 'X509_verify_cert_error_string' is deprecated (declared at /usr/include/openssl/x509.h:752)
tport_tls.c: In function 'tls_init_context':
tport_tls.c:278: warning: 'RAND_load_file' is deprecated (declared at /usr/include/openssl/rand.h:108)
tport_tls.c:301: warning: 'TLSv1_method' is deprecated (declared at /usr/include/openssl/ssl.h:1519)
tport_tls.c:303: warning: 'SSLv23_method' is deprecated (declared at /usr/include/openssl/ssl.h:1515)
tport_tls.c:305: warning: 'SSL_CTX_new' is deprecated (declared at /usr/include/openssl/ssl.h:1346)
tport_tls.c:314: warning: 'SSL_CTX_set_timeout' is deprecated (declared at /usr/include/openssl/ssl.h:1348)
tport_tls.c:318: warning: 'SSL_CTX_set_default_passwd_cb' is deprecated (declared at /usr/include/openssl/ssl.h:1472)
tport_tls.c:319: warning: 'SSL_CTX_set_default_passwd_cb_userdata' is deprecated (declared at /usr/include/openssl/ssl.h:1473)
tport_tls.c:322: warning: 'SSL_CTX_use_certificate_file' is deprecated (declared at /usr/include/openssl/ssl.h:1402)
tport_tls.c:336: warning: 'SSL_CTX_use_PrivateKey_file' is deprecated (declared at /usr/include/openssl/ssl.h:1401)
tport_tls.c:350: warning: 'SSL_CTX_check_private_key' is deprecated (declared at /usr/include/openssl/ssl.h:1475)
tport_tls.c:361: warning: 'SSL_CTX_load_verify_locations' is deprecated (declared at /usr/include/openssl/ssl.h:1572)
tport_tls.c:384: warning: 'SSL_CTX_set_verify_depth' is deprecated (declared at /usr/include/openssl/ssl.h:1460)
tport_tls.c:385: warning: 'SSL_CTX_set_verify' is deprecated (declared at /usr/include/openssl/ssl.h:1459)
tport_tls.c:387: warning: 'SSL_CTX_set_cipher_list' is deprecated (declared at /usr/include/openssl/ssl.h:1345)
tport_tls.c: In function 'tls_free':
tport_tls.c:403: warning: 'SSL_shutdown' is deprecated (declared at /usr/include/openssl/ssl.h:1532)
tport_tls.c:406: warning: 'SSL_CTX_free' is deprecated (declared at /usr/include/openssl/ssl.h:1347)
tport_tls.c:409: warning: 'BIO_free' is deprecated (declared at /usr/include/openssl/bio.h:583)
tport_tls.c: In function 'tls_get_socket':
tport_tls.c:419: warning: 'BIO_ctrl' is deprecated (declared at /usr/include/openssl/bio.h:590)
tport_tls.c: In function 'tls_init_master':
tport_tls.c:446: warning: 'RAND_pseudo_bytes' is deprecated (declared at /usr/include/openssl/rand.h:105)
tport_tls.c:448: warning: 'SSL_CTX_set_session_id_context' is deprecated (declared at /usr/include/openssl/ssl.h:1479)
tport_tls.c:453: warning: 'SSL_CTX_set_client_CA_list' is deprecated (declared at /usr/include/openssl/ssl.h:1542)
tport_tls.c:454: warning: 'SSL_load_client_CA_file' is deprecated (declared at /usr/include/openssl/ssl.h:1404)
tport_tls.c: In function 'tls_init_secondary':
tport_tls.c:495: warning: 'BIO_new_socket' is deprecated (declared at /usr/include/openssl/bio.h:675)
tport_tls.c:496: warning: 'SSL_new' is deprecated (declared at /usr/include/openssl/ssl.h:1481)
tport_tls.c:505: warning: 'SSL_set_bio' is deprecated (declared at /usr/include/openssl/ssl.h:1375)
tport_tls.c:506: warning: 'SSL_ctrl' is deprecated (declared at /usr/include/openssl/ssl.h:1496)
tport_tls.c:507: warning: 'SSL_set_ex_data' is deprecated (declared at /usr/include/openssl/ssl.h:1586)
tport_tls.c: In function 'tls_post_connection_check':
tport_tls.c:523: warning: 'SSL_get_peer_certificate' is deprecated (declared at /usr/include/openssl/ssl.h:1450)
tport_tls.c:539: warning: 'X509_get_ext_count' is deprecated (declared at /usr/include/openssl/x509.h:1144)
tport_tls.c:554: warning: 'X509_get_ext' is deprecated (declared at /usr/include/openssl/x509.h:1148)
tport_tls.c:555: warning: 'OBJ_nid2sn' is deprecated (declared at /usr/include/openssl/objects.h:1008)
tport_tls.c:555: warning: 'OBJ_obj2nid' is deprecated (declared at /usr/include/openssl/objects.h:1009)
tport_tls.c:555: warning: 'X509_EXTENSION_get_object' is deprecated (declared at /usr/include/openssl/x509.h:1185)
tport_tls.c:560: warning: 'X509V3_EXT_get' is deprecated (declared at /usr/include/openssl/x509v3.h:581)
tport_tls.c:561: warning: 'X509V3_EXT_d2i' is deprecated (declared at /usr/include/openssl/x509v3.h:585)
tport_tls.c:564: warning: 'sk_num' is deprecated (declared at /usr/include/openssl/stack.h:81)
tport_tls.c:565: warning: 'sk_value' is deprecated (declared at /usr/include/openssl/stack.h:82)
tport_tls.c:579: warning: 'X509_get_subject_name' is deprecated (declared at /usr/include/openssl/x509.h:1013)
tport_tls.c:582: warning: 'X509_NAME_get_text_by_NID' is deprecated (declared at /usr/include/openssl/x509.h:1099)
tport_tls.c:599: warning: 'SSL_get_verify_result' is deprecated (declared at /usr/include/openssl/ssl.h:1584)
tport_tls.c: In function 'tls_error':
tport_tls.c:667: warning: 'SSL_get_error' is deprecated (declared at /usr/include/openssl/ssl.h:1501)
tport_tls.c:682: warning: 'SSL_get_shutdown' is deprecated (declared at /usr/include/openssl/ssl.h:1568)
tport_tls.c: In function 'tls_read':
tport_tls.c:725: warning: 'SSL_read' is deprecated (declared at /usr/include/openssl/ssl.h:1493)
tport_tls.c: In function 'tls_pending':
tport_tls.c:741: warning: 'SSL_pending' is deprecated (declared at /usr/include/openssl/ssl.h:1368)
tport_tls.c: In function 'tls_write':
tport_tls.c:809: warning: 'SSL_write' is deprecated (declared at /usr/include/openssl/ssl.h:1495)
tport_tls.c: In function 'tls_connect':
tport_tls.c:901: warning: 'SSL_accept' is deprecated (declared at /usr/include/openssl/ssl.h:1491)
tport_tls.c:901: warning: 'SSL_connect' is deprecated (declared at /usr/include/openssl/ssl.h:1492)
tport_tls.c:902: warning: 'SSL_get_error' is deprecated (declared at /usr/include/openssl/ssl.h:1501)
tport_tls.c:956: warning: 'ERR_error_string_n' is deprecated (declared at /usr/include/openssl/err.h:280)
make[9]: *** [tport_tls.lo] Error 1
make[8]: *** [all] Error 2
Making all in nta
Making all in nth
Making all in nea
Making all in iptsec
Making all in nua
make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'.  Stop.
make[7]: *** [all-recursive] Error 1
Making all in packages
make[6]: *** [all-recursive] Error 1
make[5]: *** [all] Error 2
make[4]: *** [/Users/include/Documents/dev/Projects/WIMM/freeswitch_src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2
make[3]: *** [mod_sofia-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2

$

Any magic potion? :)

Cheers
Frank

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From jeff at jefflenk.com  Fri Aug 24 19:45:39 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 24 Aug 2012 08:45:39 -0700 (PDT)
Subject: [Freeswitch-users] Compilation error on debian
In-Reply-To: <75A08564-4DBC-410F-B679-6CF8E94FA7B9@jerris.com>
References: <50378FE5.2010608@anew.com.ve>
	<75A08564-4DBC-410F-B679-6CF8E94FA7B9@jerris.com>
Message-ID: <1345823139419-7582174.post@n2.nabble.com>

Sorry I thought this was fixed here -

SHA-1: a436a3e9624c33943a001a32aa840ca96a93f5bd

* FreeSWITCH: Fix format string error in
witch_core_session_thread_pool_worker().



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Compilation-error-on-debian-tp7582170p7582174.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From shaheryarkh at googlemail.com  Fri Aug 24 19:59:18 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Fri, 24 Aug 2012 17:59:18 +0200
Subject: [Freeswitch-users] [Freeswitch-dev] Scaled or distrbuted
	mod_conference?
In-Reply-To: <00ba01cd81e9$b5523880$1ff6a980$@207me.com>
References: <00ba01cd81e9$b5523880$1ff6a980$@207me.com>
Message-ID: <CAFZQphzxstRwgo2G=f4FMTsh46NqFu5ja7sP2XAh06Ci=+4T6w@mail.gmail.com>

hummm, interesting, i was just wondering if it is possible to make a tree
shape conferencing bridge which would work something like this.

1. We will have many freeswitch boxes, each will running a single
conference of its own up to say 50 users. These freeswitch box may be
geographically distributed as needed, (though it may cause significant
latency).

2. Then there is a freeswitch box running on top, all freeswitch boxes dial
this top level freeswitch box as normal call. On top level freeswitch these
calls will be answered and joined in a single so called top level
conference. The second leg of this call will be joined to conference
already running in lower level freeswitch boxes.

3. The main speaker / teacher of this virtual class room will just dial to
top level freeswitch and join in the top level conference..

4. For listeners / students of this virtual class to dial in and join, we
can setup an OpenSIPs / Kamailio boxes which authenticates users first then
forwards call to lower level freeswitch boxes.

5. All student calls will be mute when they join the conference. The
teacher / speaker will have a web management interface which will query
each lower freeswitch box and generate a participants list. Teacher will be
able to unmute any student to listen to their questions etc. and then mute
them again to answer the question etc.

What do you guys think? Please suggest.

Thank you.


On Fri, Aug 24, 2012 at 1:15 PM, Stephen Dame <sdame at 207me.com> wrote:

> ** **
>
> *I know a few commercial companies have scaled freeswitch to handle
> 250-500 callers in a single conference across boxes with direct audio cable
> connections in data center.*
>
> * *
>
> * I have an existing application that uses speex16(flash voip) and I can
> get 50 callers in a single conference before cpu gets to 70% on a c1.medium
> instance.   I know I can throw more hardware at the problem,  but
> interested in bridging the same conference number between multiple
> freeswitch instances and presenting single ESL notifications back to the
> existing application so all users events are seen.*
>
> * *
>
> *This is a distance learning app, and not business audio conference, so
> some latency is tolerable.   I would locate the multiple freeswitch servers
> in same zone and possibly use the new high I/O EBS instances running in SSD.
> *
>
> * *
>
> *Any thoughts on this?*
>
> * *
>
> *Regards,*
>
> *Stephen*
>
> * *
>
> * *
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From stkn at freeswitch.org  Fri Aug 24 20:10:03 2012
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Fri, 24 Aug 2012 18:10:03 +0200
Subject: [Freeswitch-users] Compilation error on debian
In-Reply-To: <1345823139419-7582174.post@n2.nabble.com>
References: <50378FE5.2010608@anew.com.ve>
	<75A08564-4DBC-410F-B679-6CF8E94FA7B9@jerris.com>
	<1345823139419-7582174.post@n2.nabble.com>
Message-ID: <5037A75B.8080107@freeswitch.org>

On 24.08.2012 17:45, Jeff Lenk wrote:
> Sorry I thought this was fixed here -
> 
> SHA-1: a436a3e9624c33943a001a32aa840ca96a93f5bd
> 
> * FreeSWITCH: Fix format string error in
> witch_core_session_thread_pool_worker().

Yes, that's fixing it.

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


From don.dawson at voice-ring.com  Fri Aug 24 20:19:44 2012
From: don.dawson at voice-ring.com (Don Dawson)
Date: Fri, 24 Aug 2012 11:19:44 -0500
Subject: [Freeswitch-users] Shared Call Appearance and Transferring
 Calls on Cisco SPA504G
In-Reply-To: <0D1C698866F66045A6201FD0F59CAC90014678A921@EX.frontier.local>
References: <0D1C698866F66045A6201FD0F59CAC90014678A921@EX.frontier.local>
Message-ID: <5037A9A0.9030604@voice-ring.com>


We have used the shared line feature for some time now.  What actually 
happens when the call is picked up by the second phone sends an INVITE 
to FS and FS connects the call to the second phone, there isn't a 
transfer (REFER) between the phones because they are sharing that call.

What we have found is this worked before version 1.2.0 and now there is 
a problem.  Could you verify what you're experiencing when you pick the 
call on phone 2, is the call drops because FS is sending BYE to the 
phone that picked up the call and the caller?  The other phones have 
their lights go green because they are sent a NOTIFY with the 
appearance-state=idle.  In the SIP trace you should see an INVITE from 
the second phone, 200 OK,  BYEs and then NOTIFYs.  Is so, then you are 
experiencing the same issue we are.

Mike


On 8/23/2012 9:33 AM, Colin Mason wrote:
>
> So I have been experimenting with a key system for an office with 3 
> Cisco SPA504G phones and FreeSWITCH 1.2.1. They require 3 shared 
> lines. I have inbound and outbound calls working properly with the 
> shared lines and the phones are configured properly with shared lines, 
> Broadcom etc. Visually everything seems to work with the line 
> notifications on the 3 phones.
>
> My internal profile has:
>
>     <param name="manage-shared-appearance" value="true"/>
>
>     <param name="manage-presence" value="true"/>
>
>     <param name="dbname" value="share_presence"/>
>
>     <param name="presence-hosts" value="$${domain}"/>
>
>     <param name="multiple-registrations" value="true"/>
>
>     <param name="force-register-domain" value="$${domain}"/>
>
>     <param name="force-subscription-domain" value="$${domain}"/>
>
>     <param name="force-register-db-domain" value="$${domain}"/>
>
>     <param name="presence-probe-on-register" value="true"/>
>
>     <param name="presence-privacy" value="$${presence_privacy}"/>
>
>     <param name="send-presence-on-register" value="true"/>
>
> I am having problems with transferring calls. If I put a call on hold 
> on phone 1 and press the line 1 button on phone 1, the call resumes 
> just fine. But if I put a call on hold on phone 1 and try to resume 
> the call on phone 2 by pressing the blinking red line 1 button, the 
> phone tries to establish a new call to the user associated with line 1 
> instead of taking (transferring) the call from phone 1 to phone 2.
>
> I was wondering if anybody could help?
>
> Colin Mason
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lists at kavun.ch  Fri Aug 24 20:43:06 2012
From: lists at kavun.ch (Emrah)
Date: Fri, 24 Aug 2012 12:43:06 -0400
Subject: [Freeswitch-users] Wideband Skype to SIP?
In-Reply-To: <50376C98.8070906@coppice.org>
References: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>
	<CAKV-jFwM7TKt6HKaKfD4HEG+9KAvDh1J=ZLP+W9dwnVQfb6_Yg@mail.gmail.com>
	<CAKV-jFy2jzH99S4u9G+KL1WkWUQPC1rUEMV7pRLcL7sL-yLUyA@mail.gmail.com>
	<93FD4ECA-6F9A-4C74-8ACE-1FBC8BF405CF@kavun.ch>
	<2B2C03B0-B514-42E9-B269-2CEE8752A4D6@kavun.ch>
	<CALXCt0oudd=jr6F2bCg5pNpeZOZj0YLfivVTK6DzOXidCTkqyw@mail.gmail.com>
	<50376C98.8070906@coppice.org>
Message-ID: <F496ACC8-BBFC-4E98-8FA7-F78CA9AD2459@kavun.ch>

For now, I'm just so super impressed on how well this is all working together.
I called the PSTN and couldn't hear any distortion coming from codec compression. It was a lot clearer than making the call from Skype on my computer directly!

Amazing work, considering upgrading to a more recent version of Skype would be fantastic. Skype may restrict access to older clients at any point.

Best,
Emrah
On Aug 24, 2012, at 7:59 AM, Steve Underwood <steveu at coppice.org> wrote:

> Hi Giovanni,
> 
> On 08/24/2012 02:30 PM, Giovanni Maruzzelli wrote:
>> Skype is native at 16 khz SL, you can't have more.
> Is that a limitation of Skype on Linux, or something? The SILK codec 
> uses a sample rate of 24ksps, so if everything in the Skype world were 
> limited to 16ksps they wouldn't be getting the full benefit of their key 
> codec.
> 
> Steve
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 onyeagbaikenna04 at gmail.com  Fri Aug 24 20:48:46 2012
From: onyeagbaikenna04 at gmail.com (onyeagbaikenna04 at gmail.com)
Date: Fri, 24 Aug 2012 16:48:46 +0000
Subject: [Freeswitch-users] Problem with linking freeswitch with GUI
In-Reply-To: <CAKzWOxVM2XGehpXYzBzjTGEzicsGovoj7k_A1gVuDttCng=5YQ@mail.gmail.com>
References: <CAHibNNek8QZ-J+GzuyxNHhnoJY5XFScW2Kf0q_ES5QmT4YNQBQ@mail.gmail.com>
	<CAKzWOxVM2XGehpXYzBzjTGEzicsGovoj7k_A1gVuDttCng=5YQ@mail.gmail.com>
Message-ID: <1718334042-1345826832-cardhu_decombobulator_blackberry.rim.net-338778532-@b13.c16.bise7.blackberry>

CudaTel..
Sent from my BlackBerry smartphone from Virgin Media

-----Original Message-----
From: Michael Collins <msc at freeswitch.org>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Fri, 24 Aug 2012 08:41:00 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Problem with linking freeswitch with GUI

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 mitch.capper at gmail.com  Fri Aug 24 20:53:50 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Fri, 24 Aug 2012 09:53:50 -0700
Subject: [Freeswitch-users] Problem with linking freeswitch with GUI
In-Reply-To: <1718334042-1345826832-cardhu_decombobulator_blackberry.rim.net-338778532-@b13.c16.bise7.blackberry>
References: <CAHibNNek8QZ-J+GzuyxNHhnoJY5XFScW2Kf0q_ES5QmT4YNQBQ@mail.gmail.com>
	<CAKzWOxVM2XGehpXYzBzjTGEzicsGovoj7k_A1gVuDttCng=5YQ@mail.gmail.com>
	<1718334042-1345826832-cardhu_decombobulator_blackberry.rim.net-338778532-@b13.c16.bise7.blackberry>
Message-ID: <CALn233rY2EySP46_P+AWgUE77pp3g5VxwOoC0aaDOM2OO=YX9A@mail.gmail.com>

CudaTel is a paid device you can purchase at: /

~mitch

On Fri, Aug 24, 2012 at 9:48 AM,  <onyeagbaikenna04 at gmail.com> wrote:
> CudaTel..
> Sent from my BlackBerry smartphone from Virgin Media
>
> -----Original Message-----
> From: Michael Collins <msc at freeswitch.org>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Fri, 24 Aug 2012 08:41:00
> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Problem with linking freeswitch with GUI
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 cmason at frontiernetworks.ca  Fri Aug 24 21:18:00 2012
From: cmason at frontiernetworks.ca (Colin Mason)
Date: Fri, 24 Aug 2012 13:18:00 -0400
Subject: [Freeswitch-users] Shared Call Appearance and Transferring
 Calls on Cisco SPA504G
In-Reply-To: <5037A9A0.9030604@voice-ring.com>
References: <0D1C698866F66045A6201FD0F59CAC90014678A921@EX.frontier.local>
	<5037A9A0.9030604@voice-ring.com>
Message-ID: <0D1C698866F66045A6201FD0F59CAC90014678A9CC@EX.frontier.local>

Thanks for the replies.

I tried a few of Andrew's suggestions and I was still unable to get call pickup (transfer?) working properly.

Don, I am seeing different behavior. When I press hold on phone 1, I get an invite from phone 1 with SDP of 0.0.0.0 which puts the call on hold.

-          Phone 2's button changes to indicate that the call is on hold.

-          I then press the button to pickup the call from hold and phone 2 sends an invite to FreeSWITCH with the username associated with that button. (this may be an issue?)

-          FreeSWITCH attemps to dial the user associated with the button I pressed on phone 2.

-          Because this user already has a call active and I limit all my users to 1 concurrent call, FreeSWITCH rolls over to the second user.

-          Phone 1 and phone 2 receive a call on line 2 and line 1 never drops throughout this process.

I suspect my problem is more of a configuration issue on the phone or FreeSWITCH.

Colin

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Don Dawson
Sent: Friday, August 24, 2012 12:20 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G


We have used the shared line feature for some time now.  What actually happens when the call is picked up by the second phone sends an INVITE to FS and FS connects the call to the second phone, there isn't a transfer (REFER) between the phones because they are sharing that call.

What we have found is this worked before version 1.2.0 and now there is a problem.  Could you verify what you're experiencing when you pick the call on phone 2, is the call drops because FS is sending BYE to the phone that picked up the call and the caller?  The other phones have their lights go green because they are sent a NOTIFY with the appearance-state=idle.  In the SIP trace you should see an INVITE from the second phone, 200 OK,  BYEs and then NOTIFYs.  Is so, then you are experiencing the same issue we are.
Mike


On 8/23/2012 9:33 AM, Colin Mason wrote:
So I have been experimenting with a key system for an office with 3 Cisco SPA504G phones and FreeSWITCH 1.2.1. They require 3 shared lines. I have inbound and outbound calls working properly with the shared lines and the phones are configured properly with shared lines, Broadcom etc. Visually everything seems to work with the line notifications on the 3 phones.

My internal profile has:

    <param name="manage-shared-appearance" value="true"/>
    <param name="manage-presence" value="true"/>
    <param name="dbname" value="share_presence"/>
    <param name="presence-hosts" value="$${domain}"/>
    <param name="multiple-registrations" value="true"/>
    <param name="force-register-domain" value="$${domain}"/>
    <param name="force-subscription-domain" value="$${domain}"/>
    <param name="force-register-db-domain" value="$${domain}"/>
    <param name="presence-probe-on-register" value="true"/>
    <param name="presence-privacy" value="$${presence_privacy}"/>
    <param name="send-presence-on-register" value="true"/>

I am having problems with transferring calls. If I put a call on hold on phone 1 and press the line 1 button on phone 1, the call resud ames just fine. But if I put a call on hold on phone 1 and try to resume the call on phone 2 by pressing the blinking red line 1 button, the phone tries to establish a new call to the user associated with line 1 instead of taking (transferring) the call from phone 1 to phone 2.

I was wondering if anybody could help?

Colin Mason




_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org<mailto:consulting at freeswitch.org>

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

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

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From andrew at cassidywebservices.co.uk  Fri Aug 24 21:42:19 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Fri, 24 Aug 2012 18:42:19 +0100
Subject: [Freeswitch-users] [Freeswitch-dev] Scaled or distrbuted
	mod_conference?
In-Reply-To: <CAFZQphzxstRwgo2G=f4FMTsh46NqFu5ja7sP2XAh06Ci=+4T6w@mail.gmail.com>
References: <00ba01cd81e9$b5523880$1ff6a980$@207me.com>
	<CAFZQphzxstRwgo2G=f4FMTsh46NqFu5ja7sP2XAh06Ci=+4T6w@mail.gmail.com>
Message-ID: <CAMbW9oLoBxg2BMAstVB-qk1g=6GiUPQ06wiyWLkLv-r4NpEw1A@mail.gmail.com>

Sounds like something I was pondering before too. In theory with a little
playing around you should be able to use 'bridging' conferences on the
'low' boxes to do this.

http://wiki.freeswitch.org/wiki/Mod_conference#Syntax

>From what I can tell, when the first user dials into a low, you can make it
create a new conference call there while bridging in the top level one.

As you're writing in your own moderator controls, you're not going to have
an issue with that, in theory you can query, mute and unmute over ESL or
XML_RPC:

http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference

On 24 August 2012 16:59, Muhammad Shahzad <shaheryarkh at googlemail.com>wrote:

> hummm, interesting, i was just wondering if it is possible to make a tree
> shape conferencing bridge which would work something like this.
>
> 1. We will have many freeswitch boxes, each will running a single
> conference of its own up to say 50 users. These freeswitch box may be
> geographically distributed as needed, (though it may cause significant
> latency).
>
> 2. Then there is a freeswitch box running on top, all freeswitch boxes
> dial this top level freeswitch box as normal call. On top level freeswitch
> these calls will be answered and joined in a single so called top level
> conference. The second leg of this call will be joined to conference
> already running in lower level freeswitch boxes.
>
> 3. The main speaker / teacher of this virtual class room will just dial to
> top level freeswitch and join in the top level conference..
>
> 4. For listeners / students of this virtual class to dial in and join, we
> can setup an OpenSIPs / Kamailio boxes which authenticates users first then
> forwards call to lower level freeswitch boxes.
>
> 5. All student calls will be mute when they join the conference. The
> teacher / speaker will have a web management interface which will query
> each lower freeswitch box and generate a participants list. Teacher will be
> able to unmute any student to listen to their questions etc. and then mute
> them again to answer the question etc.
>
> What do you guys think? Please suggest.
>
> Thank you.
>
>
> On Fri, Aug 24, 2012 at 1:15 PM, Stephen Dame <sdame at 207me.com> wrote:
>
>> ** **
>>
>> *I know a few commercial companies have scaled freeswitch to handle
>> 250-500 callers in a single conference across boxes with direct audio cable
>> connections in data center.*
>>
>> * *
>>
>> * I have an existing application that uses speex16(flash voip) and I can
>> get 50 callers in a single conference before cpu gets to 70% on a c1.medium
>> instance.   I know I can throw more hardware at the problem,  but
>> interested in bridging the same conference number between multiple
>> freeswitch instances and presenting single ESL notifications back to the
>> existing application so all users events are seen.*
>>
>> * *
>>
>> *This is a distance learning app, and not business audio conference, so
>> some latency is tolerable.   I would locate the multiple freeswitch servers
>> in same zone and possibly use the new high I/O EBS instances running in SSD.
>> *
>>
>> * *
>>
>> *Any thoughts on this?*
>>
>> * *
>>
>> *Regards,*
>>
>> *Stephen*
>>
>> * *
>>
>> * *
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From andrew at cassidywebservices.co.uk  Fri Aug 24 21:45:47 2012
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Fri, 24 Aug 2012 18:45:47 +0100
Subject: [Freeswitch-users] Shared Call Appearance and Transferring
 Calls on Cisco SPA504G
In-Reply-To: <0D1C698866F66045A6201FD0F59CAC90014678A9CC@EX.frontier.local>
References: <0D1C698866F66045A6201FD0F59CAC90014678A921@EX.frontier.local>
	<5037A9A0.9030604@voice-ring.com>
	<0D1C698866F66045A6201FD0F59CAC90014678A9CC@EX.frontier.local>
Message-ID: <CAMbW9o+tk3wEJFiS3uV4MUmZjRVmM4MgKbLn3SNwEKRmWUYf3w@mail.gmail.com>

How is the SLA set up?

To my knowledge for 'true' SLA you need to set them all up using the same
SIP account. Again, not something I've tried and Don appears to be better
equipped to help you than I so may be able to clarify the correct setup for
us?

Thanks

On 24 August 2012 18:18, Colin Mason <cmason at frontiernetworks.ca> wrote:

> Thanks for the replies.****
>
> ** **
>
> I tried a few of Andrew?s suggestions and I was still unable to get call
> pickup (transfer?) working properly.****
>
> ** **
>
> Don, I am seeing different behavior. When I press hold on phone 1, I get
> an invite from phone 1 with SDP of 0.0.0.0 which puts the call on hold. **
> **
>
> **-          **Phone 2?s button changes to indicate that the call is on
> hold. ****
>
> **-          **I then press the button to pickup the call from hold and
> phone 2 sends an invite to FreeSWITCH with the username associated with
> that button. (this may be an issue?)****
>
> **-          **FreeSWITCH attemps to dial the user associated with the
> button I pressed on phone 2.****
>
> **-          **Because this user already has a call active and I limit
> all my users to 1 concurrent call, FreeSWITCH rolls over to the second user.
> ****
>
> **-          **Phone 1 and phone 2 receive a call on line 2 and line 1
> never drops throughout this process.****
>
> ** **
>
> I suspect my problem is more of a configuration issue on the phone or
> FreeSWITCH.****
>
> ** **
>
> Colin****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Don Dawson
> *Sent:* Friday, August 24, 2012 12:20 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Shared Call Appearance and Transferring
> Calls on Cisco SPA504G****
>
> ** **
>
>
> We have used the shared line feature for some time now.  What actually
> happens when the call is picked up by the second phone sends an INVITE to
> FS and FS connects the call to the second phone, there isn?t a transfer
> (REFER) between the phones because they are sharing that call.****
>
>  ****
>
> What we have found is this worked before version 1.2.0 and now there is a
> problem.  Could you verify what you?re experiencing when you pick the call
> on phone 2, is the call drops because FS is sending BYE to the phone that
> picked up the call and the caller?  The other phones have their lights go
> green because they are sent a NOTIFY with the appearance-state=idle.  In
> the SIP trace you should see an INVITE from the second phone, 200 OK,  BYEs
> and then NOTIFYs.  Is so, then you are experiencing the same issue we are.
> ****
>
> Mike
>
>
> On 8/23/2012 9:33 AM, Colin Mason wrote:****
>
> So I have been experimenting with a key system for an office with 3 Cisco
> SPA504G phones and FreeSWITCH 1.2.1. They require 3 shared lines. I have
> inbound and outbound calls working properly with the shared lines and the
> phones are configured properly with shared lines, Broadcom etc. Visually
> everything seems to work with the line notifications on the 3 phones.****
>
>  ****
>
> My internal profile has:****
>
>  ****
>
>     <param name="manage-shared-appearance" value="true"/>****
>
>     <param name="manage-presence" value="true"/>****
>
>     <param name="dbname" value="share_presence"/>****
>
>     <param name="presence-hosts" value="$${domain}"/>****
>
>     <param name="multiple-registrations" value="true"/>****
>
>     <param name="force-register-domain" value="$${domain}"/>****
>
>     <param name="force-subscription-domain" value="$${domain}"/>****
>
>     <param name="force-register-db-domain" value="$${domain}"/>****
>
>     <param name="presence-probe-on-register" value="true"/>****
>
>     <param name="presence-privacy" value="$${presence_privacy}"/>****
>
>     <param name="send-presence-on-register" value="true"/>****
>
>  ****
>
> I am having problems with transferring calls. If I put a call on hold on
> phone 1 and press the line 1 button on phone 1, the call resud ames just
> fine. But if I put a call on hold on phone 1 and try to resume the call on
> phone 2 by pressing the blinking red line 1 button, the phone tries to
> establish a new call to the user associated with line 1 instead of taking
> (transferring) the call from phone 1 to phone 2.****
>
>  ****
>
> I was wondering if anybody could help?****
>
>  ****
>
> Colin Mason****
>
>
>
>
> ****
>
> _________________________________________________________________________****
>
> Professional FreeSWITCH Consulting Services:****
>
> consulting at freeswitch.org****
>
> http://www.freeswitchsolutions.com****
>
> ** **
>
> ****
>
> ****
>
> ** **
>
> Official FreeSWITCH Sites****
>
> http://www.freeswitch.org****
>
> http://wiki.freeswitch.org****
>
> http://www.cluecon.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****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From cmason at frontiernetworks.ca  Fri Aug 24 21:52:36 2012
From: cmason at frontiernetworks.ca (Colin Mason)
Date: Fri, 24 Aug 2012 13:52:36 -0400
Subject: [Freeswitch-users] Shared Call Appearance and Transferring
 Calls on Cisco SPA504G
In-Reply-To: <CAMbW9o+tk3wEJFiS3uV4MUmZjRVmM4MgKbLn3SNwEKRmWUYf3w@mail.gmail.com>
References: <0D1C698866F66045A6201FD0F59CAC90014678A921@EX.frontier.local>
	<5037A9A0.9030604@voice-ring.com>
	<0D1C698866F66045A6201FD0F59CAC90014678A9CC@EX.frontier.local>
	<CAMbW9o+tk3wEJFiS3uV4MUmZjRVmM4MgKbLn3SNwEKRmWUYf3w@mail.gmail.com>
Message-ID: <0D1C698866F66045A6201FD0F59CAC90014678A9D0@EX.frontier.local>

Each phone is setup like:

Button 1: user_1
Button 2: user_2
Button 3: user_3

So for the 3 phones with 3 lines each there are 3 SIP accounts in total, not 9. I believe it's setup properly

Colin

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andrew Cassidy
Sent: Friday, August 24, 2012 1:46 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G

How is the SLA set up?

To my knowledge for 'true' SLA you need to set them all up using the same SIP account. Again, not something I've tried and Don appears to be better equipped to help you than I so may be able to clarify the correct setup for us?

Thanks
On 24 August 2012 18:18, Colin Mason <cmason at frontiernetworks.ca<mailto:cmason at frontiernetworks.ca>> wrote:
Thanks for the replies.

I tried a few of Andrew's suggestions and I was still unable to get call pickup (transfer?) working properly.

Don, I am seeing different behavior. When I press hold on phone 1, I get an invite from phone 1 with SDP of 0.0.0.0 which puts the call on hold.

-          Phone 2's button changes to indicate that the call is on hold.

-          I then press the button to pickup the call from hold and phone 2 sends an invite to FreeSWITCH with the username associated with that button. (this may be an issue?)

-          FreeSWITCH attemps to dial the user associated with the button I pressed on phone 2.

-          Because this user already has a call active and I limit all my users to 1 concurrent call, FreeSWITCH rolls over to the second user.

-          Phone 1 and phone 2 receive a call on line 2 and line 1 never drops throughout this process.

I suspect my problem is more of a configuration issue on the phone or FreeSWITCH.

Colin

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 Don Dawson
Sent: Friday, August 24, 2012 12:20 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G


We have used the shared line feature for some time now.  What actually happens when the call is picked up by the second phone sends an INVITE to FS and FS connects the call to the second phone, there isn't a transfer (REFER) between the phones because they are sharing that call.

What we have found is this worked before version 1.2.0 and now there is a problem.  Could you verify what you're experiencing when you pick the call on phone 2, is the call drops because FS is sending BYE to the phone that picked up the call and the caller?  The other phones have their lights go green because they are sent a NOTIFY with the appearance-state=idle.  In the SIP trace you should see an INVITE from the second phone, 200 OK,  BYEs and then NOTIFYs.  Is so, then you are experiencing the same issue we are.
Mike


On 8/23/2012 9:33 AM, Colin Mason wrote:
So I have been experimenting with a key system for an office with 3 Cisco SPA504G phones and FreeSWITCH 1.2.1. They require 3 shared lines. I have inbound and outbound calls working properly with the shared lines and the phones are configured properly with shared lines, Broadcom etc. Visually everything seems to work with the line notifications on the 3 phones.

My internal profile has:

    <param name="manage-shared-appearance" value="true"/>
    <param name="manage-presence" value="true"/>
    <param name="dbname" value="share_presence"/>
    <param name="presence-hosts" value="$${domain}"/>
    <param name="multiple-registrations" value="true"/>
    <param name="force-register-domain" value="$${domain}"/>
    <param name="force-subscription-domain" value="$${domain}"/>
    <param name="force-register-db-domain" value="$${domain}"/>
    <param name="presence-probe-on-register" value="true"/>
    <param name="presence-privacy" value="$${presence_privacy}"/>
    <param name="send-presence-on-register" value="true"/>

I am having problems with transferring calls. If I put a call on hold on phone 1 and press the line 1 button on phone 1, the call resud ames just fine. But if I put a call on hold on phone 1 and try to resume the call on phone 2 by pressing the blinking red line 1 button, the phone tries to establish a new call to the user associated with line 1 instead of taking (transferring) the call from phone 1 to phone 2.

I was wondering if anybody could help?

Colin Mason



_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org<mailto:consulting at freeswitch.org>

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>

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http://www.freeswitch.org


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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



--
Andrew Cassidy BSc (Hons) MBCS SSCA
Managing Director
[http://c1170247.r47.cf3.rackcdn.com/emailsig.png]

T<mailto:info at cassidywebservices.co.uk> 03300 100 960  F<mailto:info at cassidywebservices.co.uk> 03300 100 961
E<mailto:info at cassidywebservices.co.uk> andrew at cassidywebservices.co.uk<mailto:andrew at cassidywebservices.co.uk>
W<mailto:info at cassidywebservices.co.uk> www.cassidywebservices.co.uk<http://www.cassidywebservices.co.uk>

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From onyeagbaikenna04 at gmail.com  Fri Aug 24 22:05:05 2012
From: onyeagbaikenna04 at gmail.com (onyeagbaikenna04 at gmail.com)
Date: Fri, 24 Aug 2012 18:05:05 +0000
Subject: [Freeswitch-users] Problem with linking freeswitch with GUI
In-Reply-To: <CAKzWOxVM2XGehpXYzBzjTGEzicsGovoj7k_A1gVuDttCng=5YQ@mail.gmail.com>
References: <CAHibNNek8QZ-J+GzuyxNHhnoJY5XFScW2Kf0q_ES5QmT4YNQBQ@mail.gmail.com>
	<CAKzWOxVM2XGehpXYzBzjTGEzicsGovoj7k_A1gVuDttCng=5YQ@mail.gmail.com>
Message-ID: <322797225-1345831409-cardhu_decombobulator_blackberry.rim.net-1976785266-@b13.c16.bise7.blackberry>

I've got various GUI, which is easier n better to install on fedora.. I have got limited time pls. Thanks for your anticipated response
Sent from my BlackBerry smartphone from Virgin Media

-----Original Message-----
From: Michael Collins <msc at freeswitch.org>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Fri, 24 Aug 2012 08:41:00 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Problem with linking freeswitch with GUI

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 gmaruzz at gmail.com  Fri Aug 24 22:26:22 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 24 Aug 2012 20:26:22 +0200
Subject: [Freeswitch-users] Wideband Skype to SIP?
In-Reply-To: <50376C98.8070906@coppice.org>
References: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>
	<CAKV-jFwM7TKt6HKaKfD4HEG+9KAvDh1J=ZLP+W9dwnVQfb6_Yg@mail.gmail.com>
	<CAKV-jFy2jzH99S4u9G+KL1WkWUQPC1rUEMV7pRLcL7sL-yLUyA@mail.gmail.com>
	<93FD4ECA-6F9A-4C74-8ACE-1FBC8BF405CF@kavun.ch>
	<2B2C03B0-B514-42E9-B269-2CEE8752A4D6@kavun.ch>
	<CALXCt0oudd=jr6F2bCg5pNpeZOZj0YLfivVTK6DzOXidCTkqyw@mail.gmail.com>
	<50376C98.8070906@coppice.org>
Message-ID: <CALXCt0qOTzePSw7Vhc8uRSfeL7Ghu3og9Rmv=ojb7k6HapJZ-g@mail.gmail.com>

ciao Steve,
the Skype client exposes through API a tcp stream that is SL 16. No
way to change it.
This probably comes legacy from the times before Silk.
-giovanni

On 8/24/12, Steve Underwood <steveu at coppice.org> wrote:
> Hi Giovanni,
>
> On 08/24/2012 02:30 PM, Giovanni Maruzzelli wrote:
>> Skype is native at 16 khz SL, you can't have more.
> Is that a limitation of Skype on Linux, or something? The SILK codec
> uses a sample rate of 24ksps, so if everything in the Skype world were
> limited to 16ksps they wouldn't be getting the full benefit of their key
> codec.
>
> Steve
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 chavpaskov at shaw.ca  Fri Aug 24 23:32:17 2012
From: chavpaskov at shaw.ca (Tchavdar Paskov)
Date: Fri, 24 Aug 2012 12:32:17 -0700
Subject: [Freeswitch-users] FreeSWITCH Project Looking For Volunteers:
	Web	Site Dev/Maint
In-Reply-To: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
Message-ID: <fba5f967342.50377451@shaw.ca>

Ready to Help if you need.
Word Press
Apache/Lamp
Regards
Chav

----- Original Message -----
From: Michael Collins <msc at freeswitch.org>
Date: Monday, August 20, 2012 14:46
Subject: [Freeswitch-users] FreeSWITCH?Project Looking For Volunteers: Web	Site Dev/Maint
To: freeswitch-users at lists.freeswitch.org, freeswitch-dev at lists.freeswitch.org

> Hello all,
> 
> I need to refresh my list of people who have Web dev?skills and 
> are willing
> to assist with things like Web site design, programming, and 
> maintenance.If you have any of the following skills, or know 
> someone who does and who
> is willing to donate a few hours, please email me offlist:
> 
> HTML5
> CSS2/3
> Javascript/JQuery
> Drupal/PHP
> Wordpress/PHP
> Django/Python
> Web design/graphics
> Apache administration in LAMP environment
> 
> Please note that not every single skill is currently in demand; 
> we are
> trying to get a feel for who knows what and hopefully anticipate 
> futureneeds.
> 
> Thanks!
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
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From don.dawson at voice-ring.com  Fri Aug 24 23:39:40 2012
From: don.dawson at voice-ring.com (Don Dawson)
Date: Fri, 24 Aug 2012 14:39:40 -0500
Subject: [Freeswitch-users] Shared Call Appearance and Transferring
 Calls on Cisco SPA504G
In-Reply-To: <0D1C698866F66045A6201FD0F59CAC90014678A9D0@EX.frontier.local>
References: <0D1C698866F66045A6201FD0F59CAC90014678A921@EX.frontier.local>
	<5037A9A0.9030604@voice-ring.com>
	<0D1C698866F66045A6201FD0F59CAC90014678A9CC@EX.frontier.local>
	<CAMbW9o+tk3wEJFiS3uV4MUmZjRVmM4MgKbLn3SNwEKRmWUYf3w@mail.gmail.com>
	<0D1C698866F66045A6201FD0F59CAC90014678A9D0@EX.frontier.local>
Message-ID: <5037D87C.2040305@voice-ring.com>


I am concerned that FS is trying to INVITE the calling user, ours 
doesn't do that.  Is it possible to send us a pcap trace of this?

In the mean time, here is how our SPA phones are configured. We are 
sharing 3 lines on multiple phones.  The following is the same on all 
phones.  If you want, you can configure the 4^th button as a private 
line (which isn't included here).  The 3 shared extensions are 121, 122 
and 123.

   <CTI_Enable>no</CTI_Enable>

<Dialog_SDP_Enable>yes</Dialog_SDP_Enable>

<Linksys_Key_System>no</Linksys_Key_System>

<Share_Call_Appearance_1_>shared</Share_Call_Appearance_1_>

<Share_Call_Appearance_2_>shared</Share_Call_Appearance_2_>

<Share_Call_Appearance_3_>shared</Share_Call_Appearance_3_>

   <Extension_1_>1</Extension_1_>

   <Extension_2_>2</Extension_2_>

   <Extension_3_>3</Extension_3_>

   <Short_Name_1_>Line 1</Short_Name_1_>

   <Short_Name_2_>Line 2</Short_Name_2_>

   <Short_Name_3_>Line 3</Short_Name_3_>

<Share_Ext_1_>shared</Share_Ext_1_>

<Share_Ext_2_>shared</Share_Ext_2_>

<Share_Ext_3_>shared</Share_Ext_3_>

<Shared_User_ID_1_>121</Shared_User_ID_1_>

<Shared_User_ID_2_>122</Shared_User_ID_2_>

<Shared_User_ID_3_>123</Shared_User_ID_3_>

   <Display_Name_1_>Line 1</Display_Name_1_>

   <Display_Name_2_>Line 2</Display_Name_2_>

   <Display_Name_3_>Line 3</Display_Name_3_>

   <User_ID_1_>121</User_ID_1_>

   <User_ID_2_>122</User_ID_2_>

   <User_ID_3_>123</User_ID_3_>

The FS profile

     <param name="manage-shared-appearance" value="true"/>

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

     <param name="dbname" value="share_presence"/>

     <param name="presence-hosts" value="_DISABLED_"/>

     <param name="multiple-registrations" value="true"/>

     <param name="send-presence-on-register" value="true"/>





On 8/24/2012 12:52 PM, Colin Mason wrote:
>
> Each phone is setup like:
>
> Button 1: user_1
>
> Button 2: user_2
>
> Button 3: user_3
>
> So for the 3 phones with 3 lines each there are 3 SIP accounts in 
> total, not 9. I believe it's setup properly
>
> Colin
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Andrew Cassidy
> *Sent:* Friday, August 24, 2012 1:46 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Shared Call Appearance and 
> Transferring Calls on Cisco SPA504G
>
> How is the SLA set up?
>
> To my knowledge for 'true' SLA you need to set them all up using the 
> same SIP account. Again, not something I've tried and Don appears to 
> be better equipped to help you than I so may be able to clarify the 
> correct setup for us?
>
> Thanks
>
> On 24 August 2012 18:18, Colin Mason <cmason at frontiernetworks.ca 
> <mailto:cmason at frontiernetworks.ca>> wrote:
>
> Thanks for the replies.
>
> I tried a few of Andrew's suggestions and I was still unable to get 
> call pickup (transfer?) working properly.
>
> Don, I am seeing different behavior. When I press hold on phone 1, I 
> get an invite from phone 1 with SDP of 0.0.0.0 which puts the call on 
> hold.
>
> -Phone 2's button changes to indicate that the call is on hold.
>
> -I then press the button to pickup the call from hold and phone 2 
> sends an invite to FreeSWITCH with the username associated with that 
> button. (this may be an issue?)
>
> -FreeSWITCH attemps to dial the user associated with the button I 
> pressed on phone 2.
>
> -Because this user already has a call active and I limit all my users 
> to 1 concurrent call, FreeSWITCH rolls over to the second user.
>
> -Phone 1 and phone 2 receive a call on line 2 and line 1 never drops 
> throughout this process.
>
> I suspect my problem is more of a configuration issue on the phone or 
> FreeSWITCH.
>
> Colin
>
> *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 
> *Don Dawson
> *Sent:* Friday, August 24, 2012 12:20 PM
> *To:* freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Shared Call Appearance and 
> Transferring Calls on Cisco SPA504G
>
>
> We have used the shared line feature for some time now.  What actually 
> happens when the call is picked up by the second phone sends an INVITE 
> to FS and FS connects the call to the second phone, there isn't a 
> transfer (REFER) between the phones because they are sharing that call.
>
> What we have found is this worked before version 1.2.0 and now there 
> is a problem.  Could you verify what you're experiencing when you pick 
> the call on phone 2, is the call drops because FS is sending BYE to 
> the phone that picked up the call and the caller?  The other phones 
> have their lights go green because they are sent a NOTIFY with the 
> appearance-state=idle.  In the SIP trace you should see an INVITE from 
> the second phone, 200 OK,  BYEs and then NOTIFYs.  Is so, then you are 
> experiencing the same issue we are.
>
> Mike
>
>
> On 8/23/2012 9:33 AM, Colin Mason wrote:
>
>     So I have been experimenting with a key system for an office with
>     3 Cisco SPA504G phones and FreeSWITCH 1.2.1. They require 3 shared
>     lines. I have inbound and outbound calls working properly with the
>     shared lines and the phones are configured properly with shared
>     lines, Broadcom etc. Visually everything seems to work with the
>     line notifications on the 3 phones.
>
>     My internal profile has:
>
>     <param name="manage-shared-appearance" value="true"/>
>
>     <param name="manage-presence" value="true"/>
>
>     <param name="dbname" value="share_presence"/>
>
>     <param name="presence-hosts" value="$${domain}"/>
>
>     <param name="multiple-registrations" value="true"/>
>
>     <param name="force-register-domain" value="$${domain}"/>
>
>     <param name="force-subscription-domain" value="$${domain}"/>
>
>     <param name="force-register-db-domain" value="$${domain}"/>
>
>     <param name="presence-probe-on-register" value="true"/>
>
>     <param name="presence-privacy" value="$${presence_privacy}"/>
>
>     <param name="send-presence-on-register" value="true"/>
>
>     I am having problems with transferring calls. If I put a call on
>     hold on phone 1 and press the line 1 button on phone 1, the call
>     resud ames just fine. But if I put a call on hold on phone 1 and
>     try to resume the call on phone 2 by pressing the blinking red
>     line 1 button, the phone tries to establish a new call to the user
>     associated with line 1 instead of taking (transferring) the call
>     from phone 1 to phone 2.
>
>     I was wondering if anybody could help?
>
>     Colin Mason
>
>
>
>     _________________________________________________________________________
>
>     Professional FreeSWITCH Consulting Services:
>
>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>
>     http://www.freeswitchsolutions.com
>
>       
>
>     
>
>     
>
>       
>
>     Official FreeSWITCH Sites
>
>     http://www.freeswitch.org
>
>     http://wiki.freeswitch.org
>
>     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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> 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
>
>
>
> -- 
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>
> Managing Director
>
> *T <mailto:info at cassidywebservices.co.uk> *03300 100 960 *F 
> <mailto:info at cassidywebservices.co.uk> *03300 100 961
>
> *E <mailto:info at cassidywebservices.co.uk> 
> *andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk>
>
> *W <mailto:info at cassidywebservices.co.uk> 
> *www.cassidywebservices.co.uk <http://www.cassidywebservices.co.uk>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mike.burlingame at me.com  Fri Aug 24 23:48:26 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 24 Aug 2012 12:48:26 -0700
Subject: [Freeswitch-users] enable-100rel question
Message-ID: <03F639A3-5709-462F-884E-348F18074DBB@me.com>

I was looking to turn on 100rel however based on the below info from the wiki I am not sure I want to run the risk of it crashing FS is this information still accurate?


>From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files
enable-100rel
This enable support for 100rel (100% reliability - PRACK message as defined in RFC3262) This fixes a problem with SIP where provisional messages like "180 Ringing" are not ACK'd and therefore could be dropped over a poor connection without retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash, see FSCORE-392.
<param name="enable-100rel" value="true"/>
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From mike.burlingame at me.com  Fri Aug 24 23:53:09 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 24 Aug 2012 12:53:09 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
Message-ID: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>

We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?

A-Leg
Invite into Freeswitch
100 Trying back from FS to A-Leg
180 Ringing from FS to A-Leg
200 OK from FS to A-Leg at 15:08:14.638799
Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515
100 Giving a try form A-Leg to FS at 15:08:14.749757
491 From A-Leg to FS at 15:08:14.780968
ACK from FS to A-Leg at 15:08:14.781102
ACK from FS to A-Leg at 15:08:14.797143
BYE from A-LEG to FS

B-Leg
Invite from Freeswitch to B-Leg
100 Giving a try from B-Leg
180 Ringing from B-Leg
200 OK  from B-Leg at 15:08:14.635670
ACK from FS to B-Leg  at 15:08:14.637044
Invite from B-Leg to FS at 15:08:14.748623
100 Trying from FS to B-Leg at 15:08:14.748954
491 from FS to B-Leg at 15:08:14.782169
ACK from B-Leg to FS at 15:08:14.782372
BYE from FS to B-Leg
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From govoiper at gmail.com  Sat Aug 25 01:21:19 2012
From: govoiper at gmail.com (SamyGo)
Date: Sat, 25 Aug 2012 02:21:19 +0500
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
References: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
Message-ID: <CAJUJwtjYmqUZDMM+pPcekYmxeV9=qBrwM4mLtZJYqmOe-FyrmQ@mail.gmail.com>

Hi,

RE*-Invite from B-Leg to FS at 15:08:14.748623*  ? Is you phone at B leg
trying to do some kind of reinvite stuff..Disable this from FS like as in
asterisk we had an option canreinvite=no likewise something in FS.

I really hope that blocking RE-INVITES from FS will block new invites from
the B leg and hence you'r call will stay established.


BTW since both legs are in race condition,which do you think wins the race?
j/k :P

Thanks
Sammy



On Sat, Aug 25, 2012 at 12:53 AM, Mike Burlingame <mike.burlingame at me.com>wrote:

> We are seeing some instances when we send a invite from the B-Leg back to
> FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has
> not fully completed yet causing a return of a 491 from the A-Leg side
> causing the call to be disconnected. wanted to see if anyone else has seen
> something like this while running FS and if anyone had any suggestions on a
> fix?
>
> A-Leg
> Invite into Freeswitch
> 100 Trying back from FS to A-Leg
> 180 Ringing from FS to A-Leg
> 200 OK from FS to A-Leg at 15:08:14.638799
> Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515
> 100 Giving a try form A-Leg to FS at 15:08:14.749757
> 491 From A-Leg to FS at 15:08:14.780968
> ACK from FS to A-Leg at 15:08:14.781102
> ACK from FS to A-Leg at 15:08:14.797143
> BYE from A-LEG to FS
>
> B-Leg
> Invite from Freeswitch to B-Leg
> 100 Giving a try from B-Leg
> 180 Ringing from B-Leg
> 200 OK  from B-Leg at 15:08:14.635670
> ACK from FS to B-Leg  at 15:08:14.637044
> Invite from B-Leg to FS at 15:08:14.748623
> 100 Trying from FS to B-Leg at 15:08:14.748954
> 491 from FS to B-Leg at 15:08:14.782169
> ACK from B-Leg to FS at 15:08:14.782372
> BYE from FS to B-Leg
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 cmason at frontiernetworks.ca  Sat Aug 25 01:23:48 2012
From: cmason at frontiernetworks.ca (Colin Mason)
Date: Fri, 24 Aug 2012 17:23:48 -0400
Subject: [Freeswitch-users] Shared Call Appearance and Transferring
 Calls on Cisco SPA504G
In-Reply-To: <5037D87C.2040305@voice-ring.com>
References: <0D1C698866F66045A6201FD0F59CAC90014678A921@EX.frontier.local>
	<5037A9A0.9030604@voice-ring.com>
	<0D1C698866F66045A6201FD0F59CAC90014678A9CC@EX.frontier.local>
	<CAMbW9o+tk3wEJFiS3uV4MUmZjRVmM4MgKbLn3SNwEKRmWUYf3w@mail.gmail.com>
	<0D1C698866F66045A6201FD0F59CAC90014678A9D0@EX.frontier.local>
	<5037D87C.2040305@voice-ring.com>
Message-ID: <0D1C698866F66045A6201FD0F59CAC90014678AA01@EX.frontier.local>

Give me some more time and I will get you a full pcap of what is going on.

After more troubleshooting this issue I've found that I can get transferring to work on one of my SIP profiles. I have 2 profiles that phones register to. One is on the internet and one is on my LAN. I have 3 phones registering from my LAN (mpls sip profile) and 3 phones registering from the internet (internet sip profile). There are 6 SIP accounts in total.

if I call the LAN phones (mpls profile) I am able to put a call on hold and pick it up using another phone.

If I call the internet phones (internet profile) I am NOT able to put a call on hold and pick it up using another phone and the behavior I described earlier occurs.
The LAN phones and the internet phones both send an invite when trying to pickup a held call. The only difference in the invites is the domain so this should be a FreeSWITCH configuration issue:

LAN:
   INVITE sip:149.x.x.x SIP/2.0
   From: "Line 1" <sip:B105_1 at 149.x.x.x>;tag=603fbd37915ee220o0
   To: "Line 1" <sip:B105_1 at 149.x.x.x>


Internet:
   INVITE sip:192.168.30.40 SIP/2.0
   From: "Line 1" <sip:B102_1 at 192.168.30.40>;tag=1c523101af886951o0
   To: "Line 1" <sip:B102_1 at 192.168.30.40>






From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Don Dawson
Sent: Friday, August 24, 2012 3:40 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G


I am concerned that FS is trying to INVITE the calling user, ours doesn't do that.  Is it possible to send us a pcap trace of this?

In the mean time, here is how our SPA phones are configured.  We are sharing 3 lines on multiple phones.  The following is the same on all phones.  If you want, you can configure the 4th button as a private line (which isn't included here).  The 3 shared extensions are 121, 122 and 123.

  <CTI_Enable>no</CTI_Enable>
  <Dialog_SDP_Enable>yes</Dialog_SDP_Enable>
  <Linksys_Key_System>no</Linksys_Key_System>

  <Share_Call_Appearance_1_>shared</Share_Call_Appearance_1_>
  <Share_Call_Appearance_2_>shared</Share_Call_Appearance_2_>
  <Share_Call_Appearance_3_>shared</Share_Call_Appearance_3_>
  <Extension_1_>1</Extension_1_>
  <Extension_2_>2</Extension_2_>
  <Extension_3_>3</Extension_3_>
  <Short_Name_1_>Line 1</Short_Name_1_>
  <Short_Name_2_>Line 2</Short_Name_2_>
  <Short_Name_3_>Line 3</Short_Name_3_>
  <Share_Ext_1_>shared</Share_Ext_1_>
  <Share_Ext_2_>shared</Share_Ext_2_>
  <Share_Ext_3_>shared</Share_Ext_3_>
  <Shared_User_ID_1_>121</Shared_User_ID_1_>
  <Shared_User_ID_2_>122</Shared_User_ID_2_>
  <Shared_User_ID_3_>123</Shared_User_ID_3_>
  <Display_Name_1_>Line 1</Display_Name_1_>
  <Display_Name_2_>Line 2</Display_Name_2_>
  <Display_Name_3_>Line 3</Display_Name_3_>
  <User_ID_1_>121</User_ID_1_>
  <User_ID_2_>122</User_ID_2_>
  <User_ID_3_>123</User_ID_3_>

The FS profile
    <param name="manage-shared-appearance" value="true"/>
    <param name="manage-presence" value="true"/>
    <param name="dbname" value="share_presence"/>
    <param name="presence-hosts" value="_DISABLED_"/>
    <param name="multiple-registrations" value="true"/>
    <param name="send-presence-on-register" value="true"/>





On 8/24/2012 12:52 PM, Colin Mason wrote:
Each phone is setup like:

Button 1: user_1
Button 2: user_2
Button 3: user_3

So for the 3 phones with 3 lines each there are 3 SIP accounts in total, not 9. I believe it's setup properly

Colin

From: 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 Andrew Cassidy
Sent: Friday, August 24, 2012 1:46 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G

How is the SLA set up?

To my knowledge for 'true' SLA you need to set them all up using the same SIP account. Again, not something I've tried and Don appears to be better equipped to help you than I so may be able to clarify the correct setup for us?

Thanks
On 24 August 2012 18:18, Colin Mason <cmason at frontiernetworks.ca<mailto:cmason at frontiernetworks.ca>> wrote:
Thanks for the replies.

I tried a few of Andrew's suggestions and I was still unable to get call pickup (transfer?) working properly.

Don, I am seeing different behavior. When I press hold on phone 1, I get an invite from phone 1 with SDP of 0.0.0.0 which puts the call on hold.

-          Phone 2's button changes to indicate that the call is on hold.

-          I then press the button to pickup the call from hold and phone 2 sends an invite to FreeSWITCH with the username associated with that button. (this may be an issue?)

-          FreeSWITCH attemps to dial the user associated with the button I pressed on phone 2.

-          Because this user already has a call active and I limit all my users to 1 concurrent call, FreeSWITCH rolls over to the second user.

-          Phone 1 and phone 2 receive a call on line 2 and line 1 never drops throughout this process.

I suspect my problem is more of a configuration issue on the phone or FreeSWITCH.

Colin

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 Don Dawson
Sent: Friday, August 24, 2012 12:20 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G


We have used the shared line feature for some time now.  What actually happens when the call is picked up by the second phone sends an INVITE to FS and FS connects the call to the second phone, there isn't a transfer (REFER) between the phones because they are sharing that call.

What we have found is this worked before version 1.2.0 and now there is a problem.  Could you verify what you're experiencing when you pick the call on phone 2, is the call drops because FS is sending BYE to the phone that picked up the call and the caller?  The other phones have their lights go green because they are sent a NOTIFY with the appearance-state=idle.  In the SIP trace you should see an INVITE from the second phone, 200 OK,  BYEs and then NOTIFYs.  Is so, then you are experiencing the same issue we are.
Mike


On 8/23/2012 9:33 AM, Colin Mason wrote:
So I have been experimenting with a key system for an office with 3 Cisco SPA504G phones and FreeSWITCH 1.2.1. They require 3 shared lines. I have inbound and outbound calls working properly with the shared lines and the phones are configured properly with shared lines, Broadcom etc. Visually everything seems to work with the line notifications on the 3 phones.

My internal profile has:

    <param name="manage-shared-appearance" value="true"/>
    <param name="manage-presence" value="true"/>
    <param name="dbname" value="share_presence"/>
    <param name="presence-hosts" value="$${domain}"/>
    <param name="multiple-registrations" value="true"/>
    <param name="force-register-domain" value="$${domain}"/>
    <param name="force-subscription-domain" value="$${domain}"/>
    <param name="force-register-db-domain" value="$${domain}"/>
    <param name="presence-probe-on-register" value="true"/>
    <param name="presence-privacy" value="$${presence_privacy}"/>
    <param name="send-presence-on-register" value="true"/>

I am having problems with transferring calls. If I put a call on hold on phone 1 and press the line 1 button on phone 1, the call resud ames just fine. But if I put a call on hold on phone 1 and try to resume the call on phone 2 by pressing the blinking red line 1 button, the phone tries to establish a new call to the user associated with line 1 instead of taking (transferring) the call from phone 1 to phone 2.

I was wondering if anybody could help?

Colin Mason




_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org<mailto:consulting at freeswitch.org>

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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



--
Andrew Cassidy BSc (Hons) MBCS SSCA
Managing Director
[http://c1170247.r47.cf3.rackcdn.com/emailsig.png]

T<mailto:info at cassidywebservices.co.uk> 03300 100 960  F<mailto:info at cassidywebservices.co.uk> 03300 100 961
E<mailto:info at cassidywebservices.co.uk> andrew at cassidywebservices.co.uk<mailto:andrew at cassidywebservices.co.uk>
W<mailto:info at cassidywebservices.co.uk> www.cassidywebservices.co.uk<http://www.cassidywebservices.co.uk>





_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org<mailto:consulting at freeswitch.org>

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

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

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From peter.olsson at visionutveckling.se  Sat Aug 25 01:33:11 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 24 Aug 2012 21:33:11 +0000
Subject: [Freeswitch-users] enable-100rel question
In-Reply-To: <03F639A3-5709-462F-884E-348F18074DBB@me.com>
References: <03F639A3-5709-462F-884E-348F18074DBB@me.com>
Message-ID: <1FFF97C269757C458224B7C895F35F1514E359@cantor.std.visionutv.se>

Since the ticket is closed, I guess it works just fine.
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Mike Burlingame [mike.burlingame at me.com]
Skickat: den 24 augusti 2012 21:48
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] enable-100rel question

I was looking to turn on 100rel however based on the below info from the wiki I am not sure I want to run the risk of it crashing FS is this information still accurate?


>From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files
enable-100rel
This enable support for 100rel (100% reliability - PRACK message as defined in RFC3262<http://tools.ietf.org/html/rfc3262>) This fixes a problem with SIP where provisional messages like "180 Ringing" are not ACK'd and therefore could be dropped over a poor connection without retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash, see FSCORE-392<http://jira.freeswitch.org/browse/FSCORE-392>.

<param name="enable-100rel" value="true"/>

!DSPAM:5037d87c32761841043968!


From cmason at frontiernetworks.ca  Sat Aug 25 04:13:35 2012
From: cmason at frontiernetworks.ca (Colin Mason)
Date: Fri, 24 Aug 2012 20:13:35 -0400
Subject: [Freeswitch-users] Shared Call Appearance and Transferring
 Calls on Cisco SPA504G
In-Reply-To: <0D1C698866F66045A6201FD0F59CAC90014678AA01@EX.frontier.local>
References: <0D1C698866F66045A6201FD0F59CAC90014678A921@EX.frontier.local>
	<5037A9A0.9030604@voice-ring.com>
	<0D1C698866F66045A6201FD0F59CAC90014678A9CC@EX.frontier.local>
	<CAMbW9o+tk3wEJFiS3uV4MUmZjRVmM4MgKbLn3SNwEKRmWUYf3w@mail.gmail.com>
	<0D1C698866F66045A6201FD0F59CAC90014678A9D0@EX.frontier.local>
	<5037D87C.2040305@voice-ring.com>
	<0D1C698866F66045A6201FD0F59CAC90014678AA01@EX.frontier.local>
Message-ID: <0D1C698866F66045A6201FD0F59CAC90014678AA0A@EX.frontier.local>

Here is my issue:

My $${domain} is the IP of my LAN interface (mpls sip_profile) or 192.168.30.40

When trying to pickup a call on a phone registered to the mpls sip_profile it works:

2012-08-24 19:19:39.575310 [ERR] sofia.c:8810 PICK SQL select call_id from sip_dialogs where call_info='appearance-index=1' and ((sip_from_user='B102_1' and sip_from_host='192.168.30.40') or presence_id='B102_1 at 192.168.30.40') and call_id is not null [01e47cbc-68e5-1230-1cbe-b2f5313092f2] [01e47cbc-68e5-1230-1cbe-b2f5313092f2] 1

+--------------------+---------------+---------------+----------------------+--------------------------------------+
| call_info          | sip_from_user | sip_from_host | presence_id          | call_id                              |
+--------------------+---------------+---------------+----------------------+--------------------------------------+
| appearance-index=1 | B101_1        | 192.168.30.40 | NULL                 | NULL                                 |
| appearance-index=1 | B101_1        | 149.x.x.x   | B101_1 at 149.x.x.x   | 18824d44-9b44e958 at 10.102.44.10       |
| appearance-index=1 | B102_1        | 192.168.30.40 | B102_1 at 192.168.30.40 | 84094a45-68e6-1230-1cbe-b2f5313092f2 |
+--------------------+---------------+---------------+----------------------+--------------------------------------+






When trying to pickup a call on a phone registered to the internet sip_profile it is broken because the invite sip_from_host is the IP address of my internet sip_profile so it finds nothing in the sip_dialog table because the sip_from_host it should be looking for is $${domain}

2012-08-24 17:59:54.175306 [ERR] sofia.c:8810 PICK SQL select call_id from sip_dialogs where call_info='appearance-index=1' and ((sip_from_user='B105_1' and sip_from_host='149.x.x.x') or presence_id='B105_1 at 149.x.x.x') and call_id is not null [(null)] [] 0

+--------------------+---------------+---------------+----------------------+--------------------------------------+
| call_info          | sip_from_user | sip_from_host | presence_id          | call_id                              |
+--------------------+---------------+---------------+----------------------+--------------------------------------+
| appearance-index=1 | B105_1        | 192.168.30.40 | B105_1 at 192.168.30.40 | ea837341-68e6-1230-1cbe-b2f5313092f2 |
| appearance-index=1 | B101_1        | 192.168.30.40 | NULL                 | NULL                                 |
| appearance-index=1 | B101_1        | 149.x.x.x   | B101_1 at 149.x.x.x   | 99659a3b-d369ebf6 at 10.102.44.10       |
+--------------------+---------------+---------------+----------------------+--------------------------------------+



Any tips?

Colin



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Colin Mason
Sent: Friday, August 24, 2012 5:24 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G

Give me some more time and I will get you a full pcap of what is going on.

After more troubleshooting this issue I've found that I can get transferring to work on one of my SIP profiles. I have 2 profiles that phones register to. One is on the internet and one is on my LAN. I have 3 phones registering from my LAN (mpls sip profile) and 3 phones registering from the internet (internet sip profile). There are 6 SIP accounts in total.

if I call the LAN phones (mpls profile) I am able to put a call on hold and pick it up using another phone.

If I call the internet phones (internet profile) I am NOT able to put a call on hold and pick it up using another phone and the behavior I described earlier occurs.

The LAN phones and the internet phones both send an invite when trying to pickup a held call. The only difference in the invites is the domain so this should be a FreeSWITCH configuration issue:

LAN:
   INVITE sip:149.x.x.x SIP/2.0
   From: "Line 1" <sip:B105_1 at 149.x.x.x>;tag=603fbd37915ee220o0
   To: "Line 1" <sip:B105_1 at 149.x.x.x>


Internet:
   INVITE sip:192.168.30.40 SIP/2.0
   From: "Line 1" <sip:B102_1 at 192.168.30.40>;tag=1c523101af886951o0
   To: "Line 1" <sip:B102_1 at 192.168.30.40>






From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Don Dawson
Sent: Friday, August 24, 2012 3:40 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G


I am concerned that FS is trying to INVITE the calling user, ours doesn't do that.  Is it possible to send us a pcap trace of this?

In the mean time, here is how our SPA phones are configured.  We are sharing 3 lines on multiple phones.  The following is the same on all phones.  If you want, you can configure the 4th button as a private line (which isn't included here).  The 3 shared extensions are 121, 122 and 123.

  <CTI_Enable>no</CTI_Enable>
  <Dialog_SDP_Enable>yes</Dialog_SDP_Enable>
  <Linksys_Key_System>no</Linksys_Key_System>

  <Share_Call_Appearance_1_>shared</Share_Call_Appearance_1_>
  <Share_Call_Appearance_2_>shared</Share_Call_Appearance_2_>
  <Share_Call_Appearance_3_>shared</Share_Call_Appearance_3_>
  <Extension_1_>1</Extension_1_>
  <Extension_2_>2</Extension_2_>
  <Extension_3_>3</Extension_3_>
  <Short_Name_1_>Line 1</Short_Name_1_>
  <Short_Name_2_>Line 2</Short_Name_2_>
  <Short_Name_3_>Line 3</Short_Name_3_>
  <Share_Ext_1_>shared</Share_Ext_1_>
  <Share_Ext_2_>shared</Share_Ext_2_>
  <Share_Ext_3_>shared</Share_Ext_3_>
  <Shared_User_ID_1_>121</Shared_User_ID_1_>
  <Shared_User_ID_2_>122</Shared_User_ID_2_>
  <Shared_User_ID_3_>123</Shared_User_ID_3_>
  <Display_Name_1_>Line 1</Display_Name_1_>
  <Display_Name_2_>Line 2</Display_Name_2_>
  <Display_Name_3_>Line 3</Display_Name_3_>
  <User_ID_1_>121</User_ID_1_>
  <User_ID_2_>122</User_ID_2_>
  <User_ID_3_>123</User_ID_3_>

The FS profile
    <param name="manage-shared-appearance" value="true"/>
    <param name="manage-presence" value="true"/>
    <param name="dbname" value="share_presence"/>
    <param name="presence-hosts" value="_DISABLED_"/>
    <param name="multiple-registrations" value="true"/>
    <param name="send-presence-on-register" value="true"/>





On 8/24/2012 12:52 PM, Colin Mason wrote:
Each phone is setup like:

Button 1: user_1
Button 2: user_2
Button 3: user_3

So for the 3 phones with 3 lines each there are 3 SIP accounts in total, not 9. I believe it's setup properly

Colin

From: 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 Andrew Cassidy
Sent: Friday, August 24, 2012 1:46 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G

How is the SLA set up?

To my knowledge for 'true' SLA you need to set them all up using the same SIP account. Again, not something I've tried and Don appears to be better equipped to help you than I so may be able to clarify the correct setup for us?

Thanks
On 24 August 2012 18:18, Colin Mason <cmason at frontiernetworks.ca<mailto:cmason at frontiernetworks.ca>> wrote:
Thanks for the replies.

I tried a few of Andrew's suggestions and I was still unable to get call pickup (transfer?) working properly.

Don, I am seeing different behavior. When I press hold on phone 1, I get an invite from phone 1 with SDP of 0.0.0.0 which puts the call on hold.

-          Phone 2's button changes to indicate that the call is on hold.

-          I then press the button to pickup the call from hold and phone 2 sends an invite to FreeSWITCH with the username associated with that button. (this may be an issue?)

-          FreeSWITCH attemps to dial the user associated with the button I pressed on phone 2.

-          Because this user already has a call active and I limit all my users to 1 concurrent call, FreeSWITCH rolls over to the second user.

-          Phone 1 and phone 2 receive a call on line 2 and line 1 never drops throughout this process.

I suspect my problem is more of a configuration issue on the phone or FreeSWITCH.

Colin

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 Don Dawson
Sent: Friday, August 24, 2012 12:20 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Shared Call Appearance and Transferring Calls on Cisco SPA504G


We have used the shared line feature for some time now.  What actually happens when the call is picked up by the second phone sends an INVITE to FS and FS connects the call to the second phone, there isn't a transfer (REFER) between the phones because they are sharing that call.

What we have found is this worked before version 1.2.0 and now there is a problem.  Could you verify what you're experiencing when you pick the call on phone 2, is the call drops because FS is sending BYE to the phone that picked up the call and the caller?  The other phones have their lights go green because they are sent a NOTIFY with the appearance-state=idle.  In the SIP trace you should see an INVITE from the second phone, 200 OK,  BYEs and then NOTIFYs.  Is so, then you are experiencing the same issue we are.
Mike


On 8/23/2012 9:33 AM, Colin Mason wrote:
So I have been experimenting with a key system for an office with 3 Cisco SPA504G phones and FreeSWITCH 1.2.1. They require 3 shared lines. I have inbound and outbound calls working properly with the shared lines and the phones are configured properly with shared lines, Broadcom etc. Visually everything seems to work with the line notifications on the 3 phones.

My internal profile has:

    <param name="manage-shared-appearance" value="true"/>
    <param name="manage-presence" value="true"/>
    <param name="dbname" value="share_presence"/>
    <param name="presence-hosts" value="$${domain}"/>
    <param name="multiple-registrations" value="true"/>
    <param name="force-register-domain" value="$${domain}"/>
    <param name="force-subscription-domain" value="$${domain}"/>
    <param name="force-register-db-domain" value="$${domain}"/>
    <param name="presence-probe-on-register" value="true"/>
    <param name="presence-privacy" value="$${presence_privacy}"/>
    <param name="send-presence-on-register" value="true"/>

I am having problems with transferring calls. If I put a call on hold on phone 1 and press the line 1 button on phone 1, the call resud ames just fine. But if I put a call on hold on phone 1 and try to resume the call on phone 2 by pressing the blinking red line 1 button, the phone tries to establish a new call to the user associated with line 1 instead of taking (transferring) the call from phone 1 to phone 2.

I was wondering if anybody could help?

Colin Mason



_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org<mailto:consulting at freeswitch.org>

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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



--
Andrew Cassidy BSc (Hons) MBCS SSCA
Managing Director
[http://c1170247.r47.cf3.rackcdn.com/emailsig.png]

T<mailto:info at cassidywebservices.co.uk> 03300 100 960  F<mailto:info at cassidywebservices.co.uk> 03300 100 961
E<mailto:info at cassidywebservices.co.uk> andrew at cassidywebservices.co.uk<mailto:andrew at cassidywebservices.co.uk>
W<mailto:info at cassidywebservices.co.uk> www.cassidywebservices.co.uk<http://www.cassidywebservices.co.uk>




_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org<mailto:consulting at freeswitch.org>

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

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

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From msc at freeswitch.org  Sat Aug 25 04:17:33 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 24 Aug 2012 17:17:33 -0700
Subject: [Freeswitch-users] FreeSWITCH Project Looking For Volunteers:
 Web Site Dev/Maint
In-Reply-To: <fba5f967342.50377451@shaw.ca>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
	<fba5f967342.50377451@shaw.ca>
Message-ID: <CAKzWOxXV39R6BbNvyW3u8q6u67sTjQpGdE0cBO62G08wSNsm-Q@mail.gmail.com>

Tchavar,

I'm so glad you replied! You are the first person with any Wordpress
knowledge who has contacted us. We are investigating the possibility of
migrating from Drupal to Wordpress for FreeSWITCH.org. Have you ever done
anything like that? I hear that they have tools to assist in migrating the
data over to WP. In any case, we'd love to hear your thoughts on how to
make a really cool FreeSWITCH.org site with WP. We can set up a domain
specifically for that, something like wp.freeswitch.org.

Thanks!

-Michael

On Fri, Aug 24, 2012 at 12:32 PM, Tchavdar Paskov <chavpaskov at shaw.ca>wrote:

> Ready to Help if you need.
> Word Press
> Apache/Lamp
> Regards
> Chav
>
>
> ----- Original Message -----
> From: Michael Collins <msc at freeswitch.org>
> Date: Monday, August 20, 2012 14:46
> Subject: [Freeswitch-users] FreeSWITCH Project Looking For Volunteers: Web
> Site Dev/Maint
> To: freeswitch-users at lists.freeswitch.org,
> freeswitch-dev at lists.freeswitch.org
>
> > Hello all,
> >
> > I need to refresh my list of people who have Web dev skills and
> > are willing
> > to assist with things like Web site design, programming, and
> > maintenance.If you have any of the following skills, or know
> > someone who does and who
> > is willing to donate a few hours, please email me offlist:
> >
> > HTML5
> > CSS2/3
> > Javascript/JQuery
> > Drupal/PHP
> > Wordpress/PHP
> > Django/Python
> > Web design/graphics
> > Apache administration in LAMP environment
> >
> > Please note that not every single skill is currently in demand;
> > we are
> > trying to get a feel for who knows what and hopefully anticipate
> > futureneeds.
> >
> > Thanks!
> >
> > --
> > Michael S Collins
> > Twitter: @mercutioviz
> > http://www.FreeSWITCH.org
> > http://www.ClueCon.com
> > http://www.OSTAG.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From bdfoster at endigotech.com  Sat Aug 25 04:19:32 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 24 Aug 2012 20:19:32 -0400
Subject: [Freeswitch-users] Wideband Skype to SIP?
In-Reply-To: <CALXCt0qOTzePSw7Vhc8uRSfeL7Ghu3og9Rmv=ojb7k6HapJZ-g@mail.gmail.com>
References: <DD067AD2-B3C6-467C-8610-E770A73DEA38@kavun.ch>
	<CAKV-jFwM7TKt6HKaKfD4HEG+9KAvDh1J=ZLP+W9dwnVQfb6_Yg@mail.gmail.com>
	<CAKV-jFy2jzH99S4u9G+KL1WkWUQPC1rUEMV7pRLcL7sL-yLUyA@mail.gmail.com>
	<93FD4ECA-6F9A-4C74-8ACE-1FBC8BF405CF@kavun.ch>
	<2B2C03B0-B514-42E9-B269-2CEE8752A4D6@kavun.ch>
	<CALXCt0oudd=jr6F2bCg5pNpeZOZj0YLfivVTK6DzOXidCTkqyw@mail.gmail.com>
	<50376C98.8070906@coppice.org>
	<CALXCt0qOTzePSw7Vhc8uRSfeL7Ghu3og9Rmv=ojb7k6HapJZ-g@mail.gmail.com>
Message-ID: <CAKV-jFzBj2fK0-z+kB5MWFu0-KqbFEmMHK1=RPn3HG7yhr6n_g@mail.gmail.com>

onyeagbaikenna04 at gmail.com,

Please start another thread, do not hijack another thread.

Thanks.

-BDF


On Fri, Aug 24, 2012 at 2:26 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> ciao Steve,
> the Skype client exposes through API a tcp stream that is SL 16. No
> way to change it.
> This probably comes legacy from the times before Silk.
> -giovanni
>
> On 8/24/12, Steve Underwood <steveu at coppice.org> wrote:
> > Hi Giovanni,
> >
> > On 08/24/2012 02:30 PM, Giovanni Maruzzelli wrote:
> >> Skype is native at 16 khz SL, you can't have more.
> > Is that a limitation of Skype on Linux, or something? The SILK codec
> > uses a sample rate of 24ksps, so if everything in the Skype world were
> > limited to 16ksps they wouldn't be getting the full benefit of their key
> > codec.
> >
> > Steve
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
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From bdfoster at endigotech.com  Sat Aug 25 04:21:36 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 24 Aug 2012 20:21:36 -0400
Subject: [Freeswitch-users] Voice broadcasting
In-Reply-To: <CAKzWOxVrFetN6_+pQ5U4G0vqxT-VQWpumK+tt5D5vBZK-n2Wpg@mail.gmail.com>
References: <CAAdTMa-Qz1UyH+wXKEGXRBHF=Hjh7S_jpxe_BSphXgsrYpo_aw@mail.gmail.com>
	<CC359325.3A8AE%krice@freeswitch.org>
	<CAKV-jFzQizHQAbUEQxCPdi_Wc9odhcL0j7oc9grHX=kceZ3-aQ@mail.gmail.com>
	<CAAdTMa-wwTiyZ+=caecU8mC5K8Z2A_2Atnq4yonDRz-172BU5g@mail.gmail.com>
	<CAOL0t+LYhOBid83cWPwvUEPqyE309Vm6DC7ZkoQWO42CVUp4Og@mail.gmail.com>
	<CAKV-jFykvFmKSPf1ZGL0tdth2ZJPyk2tX0J2aiWJHYW5Zg7vRA@mail.gmail.com>
	<CAPoPLj_QVqmA43PWM9rpzWrAii3k_3kp4BhYxngip+A84NbQAw@mail.gmail.com>
	<CAKzWOxU8qtor5ZkENWdncGp89YP7vFHSQJCxD03y3zK-+chM0w@mail.gmail.com>
	<CALNZuwkobxi_6pVf14HH6P7MfMTD7eFLuP4tYmrPQvH6R7JiDQ@mail.gmail.com>
	<CAOL0t+LZ2fgh9CKpM+ncg-C9c_rHK=sYinPYRq15UF7AiYShgw@mail.gmail.com>
	<CAKzWOxVrFetN6_+pQ5U4G0vqxT-VQWpumK+tt5D5vBZK-n2Wpg@mail.gmail.com>
Message-ID: <CAKV-jFx0kRPebEgd8-teXzY87m6VjGMa2qC-5tMVvsBxN3i7BQ@mail.gmail.com>

I was thinking ICTDialer was commercial. Sorry for the noise.

-BDF

On Fri, Aug 24, 2012 at 11:39 AM, Michael Collins <msc at freeswitch.org>wrote:

> Tahir,
>
> Thanks for the clarification. This is totally cool. We <3 OSS! :)
> -MC
>
>
> On Fri, Aug 24, 2012 at 5:32 AM, tahir almas <tahir at ictinnovations.com>wrote:
>
>> Last day I checked the list and posted these two comments where
>> appropriate
>>
>> The interesting thing I like to mention that  we have commercial product
>> competitive to ICTDialer,  entirely  different in design  as compared to
>> ICTDialer  and we use commercial channels to market it rather than this
>> mailing list
>>
>> ICTDialer is purely OSS contribution for community  based on Open Source
>> communication framework Plivo / freeswitch and MySQL as back-end and
>> re-known Open Source CMS  Drupal as Front-End  with no hidden design  and
>> people here still have objection on posting ?
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From msc at freeswitch.org  Sat Aug 25 04:25:19 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 24 Aug 2012 17:25:19 -0700
Subject: [Freeswitch-users] Problem with linking freeswitch with GUI
In-Reply-To: <322797225-1345831409-cardhu_decombobulator_blackberry.rim.net-1976785266-@b13.c16.bise7.blackberry>
References: <CAHibNNek8QZ-J+GzuyxNHhnoJY5XFScW2Kf0q_ES5QmT4YNQBQ@mail.gmail.com>
	<CAKzWOxVM2XGehpXYzBzjTGEzicsGovoj7k_A1gVuDttCng=5YQ@mail.gmail.com>
	<322797225-1345831409-cardhu_decombobulator_blackberry.rim.net-1976785266-@b13.c16.bise7.blackberry>
Message-ID: <CAKzWOxU5u6M40mVph_ZTaHegCXmtUtXqZwfqBa7qCjHar7+P6w@mail.gmail.com>

You have a few free choices:
FusionPBX
blue.box
FreePyBX

CudaTel is an actual appliance (hardware and software).

If you want to download something quick and get started right away then
FusionPBX is easy. I've never used FreePyBX but I've heard good things
about it.

-MC

On Fri, Aug 24, 2012 at 11:05 AM, <onyeagbaikenna04 at gmail.com> wrote:

> I've got various GUI, which is easier n better to install on fedora.. I
> have got limited time pls. Thanks for your anticipated response
> Sent from my BlackBerry smartphone from Virgin Media
>
> -----Original Message-----
> From: Michael Collins <msc at freeswitch.org>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Fri, 24 Aug 2012 08:41:00
> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Problem with linking freeswitch with GUI
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From bdfoster at endigotech.com  Sat Aug 25 04:25:30 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 24 Aug 2012 20:25:30 -0400
Subject: [Freeswitch-users] Mod Erlang Event in Freeswitch
In-Reply-To: <CALg+e6zRTn-=znDxQMhsTEojXCg+T8MhYcPHFbYSrRJirvq_vw@mail.gmail.com>
References: <CALg+e6z0=xhmohAJHEoapsP4oGgBzg1uyQgBe0NDJSeRrEa9RA@mail.gmail.com>
	<CALg+e6zRTn-=znDxQMhsTEojXCg+T8MhYcPHFbYSrRJirvq_vw@mail.gmail.com>
Message-ID: <CAKV-jFyvQkfpMqGXihfe9ENJscQXWYs0Z6C+5ixaxEwMusr7mw@mail.gmail.com>

It might be worth filing a JIRA so someone can work on this.

-BDF

On Fri, Aug 24, 2012 at 6:51 AM, Eranga Udesh <eranga.erl at gmail.com> wrote:

> To answer my own question, the issue was with the Erlang version. It seels
> Mod Erlang Event is not compatible with R15B01 version. Once I downgrade
> the module compiling Erlang version to R14B03, it works fine.
>
> Hope it helps others in the list.
>
> - Eranga
>
>
>
>
> On Thu, Aug 23, 2012 at 5:32 PM, Eranga Udesh <eranga.erl at gmail.com>wrote:
>
>> Hi,
>>
>> I'm trying mod_erlang_event module in Freeswitch. However when a call
>> comes to extension 111, the Erlang node receives RPC call correctly (my
>> config as below) but it seems Freeswitch module does not receive the
>> returned {Ref, NewPid} tuple of the Erlang function. I use Erlang R15B01
>> version.
>>
>> I also tried with Registered process configuration as in the Mod Erlang
>> Event wiki, but there also it timeout. I wonder if it is an incompatibility
>> with Erlang 15B?
>>
>> Any ideas what could be wrong?
>>
>> Config:
>>
>> <extension name="to_erlang" continue="true">
>>    <condition field="destination_number" expression="^111$" >
>>        <action application="erlang" data="fsivr:new_call ivr at linux" />
>>    </condition>
>> </extension>
>>
>> Output in freeswitch debug:
>>
>> 2012-08-23 17:06:34.057997 [DEBUG] mod_erlang_event.c:1519 enter
>> erlang_outbound_function fsivr:new_call ivr at linux
>> 2012-08-23 17:06:34.057997 [DEBUG] mod_erlang_event.c:1525 Creating new
>> listener for session
>> 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1527 Launching new
>> listener
>> 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1537 Creating new
>> spawned session for listener
>> 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1423 rpc call:
>> fsivr:new_call(Ref)
>> 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1008 Connection Open
>> 2012-08-23 17:06:39.177997 [WARNING] mod_erlang_event.c:1436 Timed out
>> when waiting for outbound pid 12.0.0 at freeswitch@linux
>> 404fc3c7-45d1-4194-ad37-1e118fdee8d0
>> 2012-08-23 17:06:39.177997 [DEBUG] switch_channel.c:2919
>> (sofia/internal/5000 at linux4) Callstate Change RINGING -> HANGUP
>>
>> Thanks in advance.
>> - Eranga
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
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transmission. If verification is required please request a hard-copy
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From bdfoster at endigotech.com  Sat Aug 25 04:44:37 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 24 Aug 2012 20:44:37 -0400
Subject: [Freeswitch-users] Problem with linking freeswitch with GUI
In-Reply-To: <CAKzWOxU5u6M40mVph_ZTaHegCXmtUtXqZwfqBa7qCjHar7+P6w@mail.gmail.com>
References: <CAHibNNek8QZ-J+GzuyxNHhnoJY5XFScW2Kf0q_ES5QmT4YNQBQ@mail.gmail.com>
	<CAKzWOxVM2XGehpXYzBzjTGEzicsGovoj7k_A1gVuDttCng=5YQ@mail.gmail.com>
	<322797225-1345831409-cardhu_decombobulator_blackberry.rim.net-1976785266-@b13.c16.bise7.blackberry>
	<CAKzWOxU5u6M40mVph_ZTaHegCXmtUtXqZwfqBa7qCjHar7+P6w@mail.gmail.com>
Message-ID: <CAKV-jFw005GmX13LtdJea=FmZahWv_h98r_mFvbbFVMFOFzhDw@mail.gmail.com>

FusionPBX has an ISO but it's Ubuntu.

- BDF

On Fri, Aug 24, 2012 at 8:25 PM, Michael Collins <msc at freeswitch.org> wrote:

> You have a few free choices:
> FusionPBX
> blue.box
> FreePyBX
>
> CudaTel is an actual appliance (hardware and software).
>
> If you want to download something quick and get started right away then
> FusionPBX is easy. I've never used FreePyBX but I've heard good things
> about it.
>
> -MC
>
>
> On Fri, Aug 24, 2012 at 11:05 AM, <onyeagbaikenna04 at gmail.com> wrote:
>
>> I've got various GUI, which is easier n better to install on fedora.. I
>> have got limited time pls. Thanks for your anticipated response
>> Sent from my BlackBerry smartphone from Virgin Media
>>
>> -----Original Message-----
>> From: Michael Collins <msc at freeswitch.org>
>> Sender: freeswitch-users-bounces at lists.freeswitch.org
>> Date: Fri, 24 Aug 2012 08:41:00
>> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
>> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] Problem with linking freeswitch with GUI
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From b2m at a-cti.com  Sat Aug 25 15:55:00 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Sat, 25 Aug 2012 17:25:00 +0530
Subject: [Freeswitch-users] Recording Volume
Message-ID: <CAP2fbbcfO8Oo_SQ4AOmUb2W3L+oTZWtpjffAustdAGNeB_dQBg@mail.gmail.com>

Guys,

Please let me know how to tune call recording volume, Its very poor to
listen my calls on any Iphone/ipad & android devices.

Thanks,
Bala
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From freeswitch-list at puzzled.xs4all.nl  Sat Aug 25 17:20:14 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Sat, 25 Aug 2012 15:20:14 +0200
Subject: [Freeswitch-users] FreeSWITCH Project Looking For Volunteers:
 Web Site Dev/Maint
In-Reply-To: <CAKzWOxXV39R6BbNvyW3u8q6u67sTjQpGdE0cBO62G08wSNsm-Q@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
	<fba5f967342.50377451@shaw.ca>
	<CAKzWOxXV39R6BbNvyW3u8q6u67sTjQpGdE0cBO62G08wSNsm-Q@mail.gmail.com>
Message-ID: <5038D10E.3060102@puzzled.xs4all.nl>

On 25-08-12 02:17, Michael Collins wrote:
> Tchavar,
>
> I'm so glad you replied! You are the first person with any Wordpress
> knowledge who has contacted us. We are investigating the possibility of
> migrating from Drupal to Wordpress for FreeSWITCH.org.

While Wordpress is an excellent solution I thought it lacked Community 
features. If Community/social platform features are required how about 
Joomla 2.5 with an extension like JomSocial or Community Builder?

http://www.jomsocial.com/
http://www.joomlapolis.com/

Regards,
Patrick (no affiliation)



From Nabble_01394 at slickdeals.endjunk.com  Sat Aug 25 17:53:44 2012
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Sat, 25 Aug 2012 06:53:44 -0700 (PDT)
Subject: [Freeswitch-users] Problem with linking freeswitch with GUI
In-Reply-To: <CAHibNNek8QZ-J+GzuyxNHhnoJY5XFScW2Kf0q_ES5QmT4YNQBQ@mail.gmail.com>
References: <CAHibNNek8QZ-J+GzuyxNHhnoJY5XFScW2Kf0q_ES5QmT4YNQBQ@mail.gmail.com>
Message-ID: <1345902823870-7582206.post@n2.nabble.com>

At one point, I had  http://www.FusionPBX.com FusionPBX  (+ FS) installed on
my Seagate DockStar unit running on an  http://openwrt.org OpenWRT  OS and
used it to learn how to program and/or configure a basic FS system for my
needs. AFAIK,  http://www.FusionPBX.com FusionPBX  is a very simple and
straightforward FS GUI package to install. I used /svn checkout
http://fusionpbx.googlecode.com/svn/trunk/fusionpbx/ to clone the whole SVN
trunk to a USB memory stick connected to one of the USB2 ports on my Seagate
DockStar unit, reconfigured /uhttpd/ (a micro WEB server) to support the
newly downloaded  http://www.FusionPBX.com FusionPBX  source from its SVN
trunk, and the system was up running with  http://www.FusionPBX.com
FusionPBX  in no time.



-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-linking-freeswitch-with-GUI-tp7582162p7582206.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Sat Aug 25 18:54:39 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sat, 25 Aug 2012 07:54:39 -0700 (PDT)
Subject: [Freeswitch-users] Recording Volume
In-Reply-To: <CAP2fbbcfO8Oo_SQ4AOmUb2W3L+oTZWtpjffAustdAGNeB_dQBg@mail.gmail.com>
References: <CAP2fbbcfO8Oo_SQ4AOmUb2W3L+oTZWtpjffAustdAGNeB_dQBg@mail.gmail.com>
Message-ID: <1345906479516-7582207.post@n2.nabble.com>

Call volume should be adjusted at the source not at the destination. You
should investigate why fs is receiving audio that is not loud enough.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Recording-Volume-tp7582204p7582207.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Sat Aug 25 22:00:15 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 25 Aug 2012 13:00:15 -0500
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
References: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
Message-ID: <CAKbxfG8M6Zipau7Xe8i0MBLMQZ-1ySKA2J9Uw8t4geA5AYbTcw@mail.gmail.com>

Wait longer before you send it?

Fs is not a proxy so you have to be more careful.
On Aug 24, 2012 2:54 PM, "Mike Burlingame" <mike.burlingame at me.com> wrote:

> We are seeing some instances when we send a invite from the B-Leg back to
> FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has
> not fully completed yet causing a return of a 491 from the A-Leg side
> causing the call to be disconnected. wanted to see if anyone else has seen
> something like this while running FS and if anyone had any suggestions on a
> fix?
>
> A-Leg
> Invite into Freeswitch
> 100 Trying back from FS to A-Leg
> 180 Ringing from FS to A-Leg
> 200 OK from FS to A-Leg at 15:08:14.638799
> Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515
> 100 Giving a try form A-Leg to FS at 15:08:14.749757
> 491 From A-Leg to FS at 15:08:14.780968
> ACK from FS to A-Leg at 15:08:14.781102
> ACK from FS to A-Leg at 15:08:14.797143
> BYE from A-LEG to FS
>
> B-Leg
> Invite from Freeswitch to B-Leg
> 100 Giving a try from B-Leg
> 180 Ringing from B-Leg
> 200 OK  from B-Leg at 15:08:14.635670
> ACK from FS to B-Leg  at 15:08:14.637044
> Invite from B-Leg to FS at 15:08:14.748623
> 100 Trying from FS to B-Leg at 15:08:14.748954
> 491 from FS to B-Leg at 15:08:14.782169
> ACK from B-Leg to FS at 15:08:14.782372
> BYE from FS to B-Leg
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Sat Aug 25 22:18:28 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 25 Aug 2012 13:18:28 -0500
Subject: [Freeswitch-users] Shared Call Appearance and Transferring
 Calls on Cisco SPA504G
In-Reply-To: <0D1C698866F66045A6201FD0F59CAC90014678AA0A@EX.frontier.local>
References: <0D1C698866F66045A6201FD0F59CAC90014678A921@EX.frontier.local>
	<5037A9A0.9030604@voice-ring.com>
	<0D1C698866F66045A6201FD0F59CAC90014678A9CC@EX.frontier.local>
	<CAMbW9o+tk3wEJFiS3uV4MUmZjRVmM4MgKbLn3SNwEKRmWUYf3w@mail.gmail.com>
	<0D1C698866F66045A6201FD0F59CAC90014678A9D0@EX.frontier.local>
	<5037D87C.2040305@voice-ring.com>
	<0D1C698866F66045A6201FD0F59CAC90014678AA01@EX.frontier.local>
	<0D1C698866F66045A6201FD0F59CAC90014678AA0A@EX.frontier.local>
Message-ID: <CAKbxfG_iY-CYwWmS0kmuiProEzJDMZAYRwfFFzHYSBvLONrLpA@mail.gmail.com>

A this should be in Jira not ml.
B it sounds like Cisco does something evil with SLA so try using the ip of
the profile as the domain across the board....
On Aug 24, 2012 7:15 PM, "Colin Mason" <cmason at frontiernetworks.ca> wrote:

> Here is my issue:****
>
> ** **
>
> My $${domain} is the IP of my LAN interface (mpls sip_profile) or
> 192.168.30.40****
>
> ** **
>
> When trying to pickup a call on a phone registered to the mpls sip_profile
> it works:****
>
> ** **
>
> 2012-08-24 19:19:39.575310 [ERR] sofia.c:8810 PICK SQL select call_id from
> sip_dialogs where call_info='appearance-index=1' and
> ((sip_from_user='B102_1' and sip_from_host='192.168.30.40') or presence_id='
> B102_1 at 192.168.30.40') and call_id is not null
> [01e47cbc-68e5-1230-1cbe-b2f5313092f2]
> [01e47cbc-68e5-1230-1cbe-b2f5313092f2] 1****
>
> ** **
>
>
> +--------------------+---------------+---------------+----------------------+--------------------------------------+
> ****
>
> | call_info          | sip_from_user | sip_from_host |
> presence_id          | call_id                              |****
>
>
> +--------------------+---------------+---------------+----------------------+--------------------------------------+
> ****
>
> | appearance-index=1 | B101_1        | 192.168.30.40 |
> NULL                 | NULL                                 |****
>
> | appearance-index=1 | B101_1        | 149.x.x.x   | B101_1 at 149.x.x.x   |
> 18824d44-9b44e958 at 10.102.44.10       |****
>
> | appearance-index=1 | B102_1        | 192.168.30.40 |
> B102_1 at 192.168.30.40 | 84094a45-68e6-1230-1cbe-b2f5313092f2 |****
>
>
> +--------------------+---------------+---------------+----------------------+--------------------------------------+
> ****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> When trying to pickup a call on a phone registered to the internet
> sip_profile it is broken because the invite sip_from_host is the IP address
> of my internet sip_profile so it finds nothing in the sip_dialog table
> because the sip_from_host it should be looking for is $${domain}****
>
> ** **
>
> 2012-08-24 17:59:54.175306 [ERR] sofia.c:8810 PICK SQL select call_id from
> sip_dialogs where call_info='appearance-index=1' and
> ((sip_from_user='B105_1' and sip_from_host='149.x.x.x') or
> presence_id='B105_1 at 149.x.x.x') and call_id is not null [(null)] [] 0****
>
> ** **
>
>
> +--------------------+---------------+---------------+----------------------+--------------------------------------+
> ****
>
> | call_info          | sip_from_user | sip_from_host |
> presence_id          | call_id                              |****
>
>
> +--------------------+---------------+---------------+----------------------+--------------------------------------+
> ****
>
> | appearance-index=1 | B105_1        | 192.168.30.40 |
> B105_1 at 192.168.30.40 | ea837341-68e6-1230-1cbe-b2f5313092f2 |****
>
> | appearance-index=1 | B101_1        | 192.168.30.40 |
> NULL                 | NULL                                 |****
>
> | appearance-index=1 | B101_1        | 149.x.x.x   | B101_1 at 149.x.x.x   |
> 99659a3b-d369ebf6 at 10.102.44.10       |****
>
>
> +--------------------+---------------+---------------+----------------------+--------------------------------------+
> ****
>
> ** **
>
> ** **
>
> ** **
>
> Any tips?****
>
> ** **
>
> Colin****
>
> ** **
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Colin Mason
> *Sent:* Friday, August 24, 2012 5:24 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Shared Call Appearance and Transferring
> Calls on Cisco SPA504G****
>
> ** **
>
> Give me some more time and I will get you a full pcap of what is going on.
> ****
>
> ** **
>
> After more troubleshooting this issue I?ve found that I can get
> transferring to work on one of my SIP profiles. I have 2 profiles that
> phones register to. One is on the internet and one is on my LAN. I have 3
> phones registering from my LAN (mpls sip profile) and 3 phones registering
> from the internet (internet sip profile). There are 6 SIP accounts in
> total. ****
>
> ** **
>
> if I call the LAN phones (mpls profile) I am able to put a call on hold
> and pick it up using another phone.****
>
> ** **
>
> If I call the internet phones (internet profile) I am NOT able to put a
> call on hold and pick it up using another phone and the behavior I
> described earlier occurs.****
>
> ** **
>
> The LAN phones and the internet phones both send an invite when trying to
> pickup a held call. The only difference in the invites is the domain so
> this should be a FreeSWITCH configuration issue:****
>
> ** **
>
> LAN:****
>
>    INVITE sip:149.x.x.x SIP/2.0****
>
>    From: "Line 1" <sip:B105_1 at 149.x.x.x>;tag=603fbd37915ee220o0****
>
>    To: "Line 1" <sip:B105_1 at 149.x.x.x>****
>
> ** **
>
> ** **
>
> Internet:****
>
>    INVITE sip:192.168.30.40 SIP/2.0****
>
>    From: "Line 1" <sip:B102_1 at 192.168.30.40>;tag=1c523101af886951o0****
>
>    To: "Line 1" <sip:B102_1 at 192.168.30.40>****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Don
> Dawson
> *Sent:* Friday, August 24, 2012 3:40 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Shared Call Appearance and Transferring
> Calls on Cisco SPA504G****
>
> ** **
>
> ** **
>
> I am concerned that FS is trying to INVITE the calling user, ours doesn?t
> do that.  Is it possible to send us a pcap trace of this?****
>
>  ****
>
> In the mean time, here is how our SPA phones are configured.  We are
> sharing 3 lines on multiple phones.  The following is the same on all
> phones.  If you want, you can configure the 4th button as a private line
> (which isn?t included here).  The 3 shared extensions are 121, 122 and 123.
> ****
>
>  ****
>
>   <CTI_Enable>no</CTI_Enable>****
>
>   <Dialog_SDP_Enable>yes</Dialog_SDP_Enable>****
>
>   <Linksys_Key_System>no</Linksys_Key_System>****
>
>  ****
>
>   <Share_Call_Appearance_1_>shared</Share_Call_Appearance_1_>****
>
>   <Share_Call_Appearance_2_>shared</Share_Call_Appearance_2_>****
>
>   <Share_Call_Appearance_3_>shared</Share_Call_Appearance_3_>****
>
>   <Extension_1_>1</Extension_1_>****
>
>   <Extension_2_>2</Extension_2_>****
>
>   <Extension_3_>3</Extension_3_>****
>
>   <Short_Name_1_>Line 1</Short_Name_1_>****
>
>   <Short_Name_2_>Line 2</Short_Name_2_>****
>
>   <Short_Name_3_>Line 3</Short_Name_3_>****
>
>   <Share_Ext_1_>shared</Share_Ext_1_>****
>
>   <Share_Ext_2_>shared</Share_Ext_2_>****
>
>   <Share_Ext_3_>shared</Share_Ext_3_>****
>
>   <Shared_User_ID_1_>121</Shared_User_ID_1_>****
>
>   <Shared_User_ID_2_>122</Shared_User_ID_2_>****
>
>   <Shared_User_ID_3_>123</Shared_User_ID_3_>****
>
>   <Display_Name_1_>Line 1</Display_Name_1_>****
>
>   <Display_Name_2_>Line 2</Display_Name_2_>****
>
>   <Display_Name_3_>Line 3</Display_Name_3_>****
>
>   <User_ID_1_>121</User_ID_1_>****
>
>   <User_ID_2_>122</User_ID_2_>****
>
>   <User_ID_3_>123</User_ID_3_>****
>
>  ****
>
> The FS profile****
>
>     <param name="manage-shared-appearance" value="true"/>****
>
>     <param name="manage-presence" value="true"/>****
>
>     <param name="dbname" value="share_presence"/>****
>
>     <param name="presence-hosts" value="_DISABLED_"/>****
>
>     <param name="multiple-registrations" value="true"/>****
>
>     <param name="send-presence-on-register" value="true"/>
>
>
>
>
>
> On 8/24/2012 12:52 PM, Colin Mason wrote:****
>
> Each phone is setup like:****
>
>  ****
>
> Button 1: user_1****
>
> Button 2: user_2****
>
> Button 3: user_3****
>
>  ****
>
> So for the 3 phones with 3 lines each there are 3 SIP accounts in total,
> not 9. I believe it?s setup properly****
>
>  ****
>
> Colin****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
> *On Behalf Of *Andrew Cassidy
> *Sent:* Friday, August 24, 2012 1:46 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Shared Call Appearance and Transferring
> Calls on Cisco SPA504G****
>
>  ****
>
> How is the SLA set up?****
>
>  ****
>
> To my knowledge for 'true' SLA you need to set them all up using the same
> SIP account. Again, not something I've tried and Don appears to be better
> equipped to help you than I so may be able to clarify the correct setup for
> us?****
>
>  ****
>
> Thanks****
>
> On 24 August 2012 18:18, Colin Mason <cmason at frontiernetworks.ca> wrote:**
> **
>
> Thanks for the replies.****
>
>  ****
>
> I tried a few of Andrew?s suggestions and I was still unable to get call
> pickup (transfer?) working properly.****
>
>  ****
>
> Don, I am seeing different behavior. When I press hold on phone 1, I get
> an invite from phone 1 with SDP of 0.0.0.0 which puts the call on hold. **
> **
>
> -          Phone 2?s button changes to indicate that the call is on hold.
> ****
>
> -          I then press the button to pickup the call from hold and phone
> 2 sends an invite to FreeSWITCH with the username associated with that
> button. (this may be an issue?)****
>
> -          FreeSWITCH attemps to dial the user associated with the button
> I pressed on phone 2.****
>
> -          Because this user already has a call active and I limit all my
> users to 1 concurrent call, FreeSWITCH rolls over to the second user.****
>
> -          Phone 1 and phone 2 receive a call on line 2 and line 1 never
> drops throughout this process.****
>
>  ****
>
> I suspect my problem is more of a configuration issue on the phone or
> FreeSWITCH.****
>
>  ****
>
> Colin****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Don Dawson
> *Sent:* Friday, August 24, 2012 12:20 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Shared Call Appearance and Transferring
> Calls on Cisco SPA504G****
>
>  ****
>
>
> We have used the shared line feature for some time now.  What actually
> happens when the call is picked up by the second phone sends an INVITE to
> FS and FS connects the call to the second phone, there isn?t a transfer
> (REFER) between the phones because they are sharing that call.****
>
>  ****
>
> What we have found is this worked before version 1.2.0 and now there is a
> problem.  Could you verify what you?re experiencing when you pick the call
> on phone 2, is the call drops because FS is sending BYE to the phone that
> picked up the call and the caller?  The other phones have their lights go
> green because they are sent a NOTIFY with the appearance-state=idle.  In
> the SIP trace you should see an INVITE from the second phone, 200 OK,  BYEs
> and then NOTIFYs.  Is so, then you are experiencing the same issue we are.
> ****
>
> Mike
>
>
> On 8/23/2012 9:33 AM, Colin Mason wrote:****
>
> So I have been experimenting with a key system for an office with 3 Cisco
> SPA504G phones and FreeSWITCH 1.2.1. They require 3 shared lines. I have
> inbound and outbound calls working properly with the shared lines and the
> phones are configured properly with shared lines, Broadcom etc. Visually
> everything seems to work with the line notifications on the 3 phones.****
>
>  ****
>
> My internal profile has:****
>
>  ****
>
>     <param name="manage-shared-appearance" value="true"/>****
>
>     <param name="manage-presence" value="true"/>****
>
>     <param name="dbname" value="share_presence"/>****
>
>     <param name="presence-hosts" value="$${domain}"/>****
>
>     <param name="multiple-registrations" value="true"/>****
>
>     <param name="force-register-domain" value="$${domain}"/>****
>
>     <param name="force-subscription-domain" value="$${domain}"/>****
>
>     <param name="force-register-db-domain" value="$${domain}"/>****
>
>     <param name="presence-probe-on-register" value="true"/>****
>
>     <param name="presence-privacy" value="$${presence_privacy}"/>****
>
>     <param name="send-presence-on-register" value="true"/>****
>
>  ****
>
> I am having problems with transferring calls. If I put a call on hold on
> phone 1 and press the line 1 button on phone 1, the call resud ames just
> fine. But if I put a call on hold on phone 1 and try to resume the call on
> phone 2 by pressing the blinking red line 1 button, the phone tries to
> establish a new call to the user associated with line 1 instead of taking
> (transferring) the call from phone 1 to phone 2.****
>
>  ****
>
> I was wondering if anybody could help?****
>
>  ****
>
> Colin Mason****
>
>
>
> ****
>
> _________________________________________________________________________****
>
> Professional FreeSWITCH Consulting Services:****
>
> consulting at freeswitch.org****
>
> http://www.freeswitchsolutions.com****
>
>  ****
>
> ****
>
> ****
>
>  ****
>
> Official FreeSWITCH Sites****
>
> http://www.freeswitch.org****
>
> http://wiki.freeswitch.org****
>
> http://www.cluecon.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****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
>
>
> ****
>
>  ****
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*****
>
> Managing Director****
>
> ****
>
>  ****
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961****
>
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk****
>
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk****
>
>  ****
>
>
>
> ****
>
> _________________________________________________________________________****
>
> Professional FreeSWITCH Consulting Services:****
>
> consulting at freeswitch.org****
>
> http://www.freeswitchsolutions.com****
>
> ** **
>
> ****
>
> ****
>
> ** **
>
> Official FreeSWITCH Sites****
>
> http://www.freeswitch.org****
>
> http://wiki.freeswitch.org****
>
> http://www.cluecon.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****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mike.burlingame at me.com  Sat Aug 25 22:22:48 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Sat, 25 Aug 2012 11:22:48 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <CAKbxfG8M6Zipau7Xe8i0MBLMQZ-1ySKA2J9Uw8t4geA5AYbTcw@mail.gmail.com>
References: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
	<CAKbxfG8M6Zipau7Xe8i0MBLMQZ-1ySKA2J9Uw8t4geA5AYbTcw@mail.gmail.com>
Message-ID: <0004540A-8EF8-49B5-BF3A-84195A2960CA@me.com>

The M6 on the other side does not have any options to delay the invite also broadsoft support is saying due to the ACK on the B-Leg (Broadsoft side) that they see no reason to delay the invite.

This only happens on 1 out of 30-45 calls and does not seem to be resource related

I will try to work with Broadsoft to see if they can delay the invite

Thanks for your feedback

Sent from my iPad

On Aug 25, 2012, at 11:00 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> Wait longer before you send it?
> 
> Fs is not a proxy so you have to be more careful. 
> 
> On Aug 24, 2012 2:54 PM, "Mike Burlingame" <mike.burlingame at me.com> wrote:
> We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
> 
> A-Leg
> Invite into Freeswitch
> 100 Trying back from FS to A-Leg
> 180 Ringing from FS to A-Leg
> 200 OK from FS to A-Leg at 15:08:14.638799
> Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515
> 100 Giving a try form A-Leg to FS at 15:08:14.749757
> 491 From A-Leg to FS at 15:08:14.780968
> ACK from FS to A-Leg at 15:08:14.781102
> ACK from FS to A-Leg at 15:08:14.797143
> BYE from A-LEG to FS
> 
> B-Leg
> Invite from Freeswitch to B-Leg
> 100 Giving a try from B-Leg
> 180 Ringing from B-Leg
> 200 OK  from B-Leg at 15:08:14.635670
> ACK from FS to B-Leg  at 15:08:14.637044
> Invite from B-Leg to FS at 15:08:14.748623
> 100 Trying from FS to B-Leg at 15:08:14.748954
> 491 from FS to B-Leg at 15:08:14.782169
> ACK from B-Leg to FS at 15:08:14.782372
> BYE from FS to B-Leg
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 onyeagbaikenna04 at gmail.com  Sat Aug 25 22:37:28 2012
From: onyeagbaikenna04 at gmail.com (onyeagbaikenna04 at gmail.com)
Date: Sat, 25 Aug 2012 18:37:28 +0000
Subject: [Freeswitch-users] Problem with linking freeswitch with GUI
In-Reply-To: <1345902823870-7582206.post@n2.nabble.com>
References: <CAHibNNek8QZ-J+GzuyxNHhnoJY5XFScW2Kf0q_ES5QmT4YNQBQ@mail.gmail.com>
	<1345902823870-7582206.post@n2.nabble.com>
Message-ID: <262109171-1345919753-cardhu_decombobulator_blackberry.rim.net-1468029409-@b13.c16.bise7.blackberry>

Thanks for ur reply, I've got freeswitch already installed, just to be sure, am I goin to install the GUI as a seperate installation from d freeswitch?
Sent from my BlackBerry smartphone from Virgin Media

-----Original Message-----
From: mazilo <Nabble_01394 at slickdeals.endjunk.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Sat, 25 Aug 2012 06:53:44 
To: <freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Problem with linking freeswitch with GUI

At one point, I had  http://www.FusionPBX.com FusionPBX  (+ FS) installed on
my Seagate DockStar unit running on an  http://openwrt.org OpenWRT  OS and
used it to learn how to program and/or configure a basic FS system for my
needs. AFAIK,  http://www.FusionPBX.com FusionPBX  is a very simple and
straightforward FS GUI package to install. I used /svn checkout
http://fusionpbx.googlecode.com/svn/trunk/fusionpbx/ to clone the whole SVN
trunk to a USB memory stick connected to one of the USB2 ports on my Seagate
DockStar unit, reconfigured /uhttpd/ (a micro WEB server) to support the
newly downloaded  http://www.FusionPBX.com FusionPBX  source from its SVN
trunk, and the system was up running with  http://www.FusionPBX.com
FusionPBX  in no time.



-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-linking-freeswitch-with-GUI-tp7582162p7582206.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 bdfoster at endigotech.com  Sat Aug 25 23:52:47 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 25 Aug 2012 15:52:47 -0400
Subject: [Freeswitch-users] Problem with linking freeswitch with GUI
In-Reply-To: <262109171-1345919753-cardhu_decombobulator_blackberry.rim.net-1468029409-@b13.c16.bise7.blackberry>
References: <CAHibNNek8QZ-J+GzuyxNHhnoJY5XFScW2Kf0q_ES5QmT4YNQBQ@mail.gmail.com>
	<1345902823870-7582206.post@n2.nabble.com>
	<262109171-1345919753-cardhu_decombobulator_blackberry.rim.net-1468029409-@b13.c16.bise7.blackberry>
Message-ID: <CAKV-jFwUy__vV0nVreV3zn34qtM+XnBFKfNHviGcUcRXCgE8jg@mail.gmail.com>

In short, yes.

FusionPBX interfaces with FS via the event socket as well as through
editing the actual XML that FreeSWITCH reads. It's completely seperate from
FS, and really FS doesn't care what you are using. All FusionPBX is doing
is manipulating FS for you to make things easier to manage.

There are many guides on how to install FusionPBX. Your best bet is to
start with http://fusionpbx.com and http://wiki.fusionpbx.com and get the
low down. It's not difficult, and if you're on Debian, there are a few
scripts out there that will install fusionpbx and freeswitch for you. Take
a look at soapee's script here:
http://wiki.fusionpbx.com/index.php/Easy_FusionPBX , or my script:
http://github.com/bdfoster/fs-debian-installer . Mine is not as polished as
soapee's script.

- BDF

On Sat, Aug 25, 2012 at 2:37 PM, <onyeagbaikenna04 at gmail.com> wrote:

> Thanks for ur reply, I've got freeswitch already installed, just to be
> sure, am I goin to install the GUI as a seperate installation from d
> freeswitch?
> Sent from my BlackBerry smartphone from Virgin Media
>
> -----Original Message-----
> From: mazilo <Nabble_01394 at slickdeals.endjunk.com>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Sat, 25 Aug 2012 06:53:44
> To: <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Problem with linking freeswitch with GUI
>
> At one point, I had  http://www.FusionPBX.com FusionPBX  (+ FS) installed
> on
> my Seagate DockStar unit running on an  http://openwrt.org OpenWRT  OS and
> used it to learn how to program and/or configure a basic FS system for my
> needs. AFAIK,  http://www.FusionPBX.com FusionPBX  is a very simple and
> straightforward FS GUI package to install. I used /svn checkout
> http://fusionpbx.googlecode.com/svn/trunk/fusionpbx/ to clone the whole
> SVN
> trunk to a USB memory stick connected to one of the USB2 ports on my
> Seagate
> DockStar unit, reconfigured /uhttpd/ (a micro WEB server) to support the
> newly downloaded  http://www.FusionPBX.com FusionPBX  source from its SVN
> trunk, and the system was up running with  http://www.FusionPBX.com
> FusionPBX  in no time.
>
>
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
> Watts of electricity.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Problem-with-linking-freeswitch-with-GUI-tp7582162p7582206.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
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From avi at avimarcus.net  Sun Aug 26 00:36:49 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 25 Aug 2012 23:36:49 +0300
Subject: [Freeswitch-users] Subscribe to sofia:: events?
Message-ID: <CALNZuw=z+oqK61e_Gb6q5eM3dGN_PJt9PdAgb8YGHrVAqcuKhw@mail.gmail.com>

I seem to be unable to subscribe to sofia events.
Subscribing to CUSTOM doesn't show them. Subscribing to sofia::register
responds: "-ERR no keywords supplied"
Only subscribing to ALL shows them (with type CUSTOM).

Am I doing something wrong? Is this a bug?

-Avi
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From dujinfang at gmail.com  Sun Aug 26 02:28:22 2012
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 26 Aug 2012 06:28:22 +0800
Subject: [Freeswitch-users] Mod Erlang Event in Freeswitch
In-Reply-To: <CAKV-jFyvQkfpMqGXihfe9ENJscQXWYs0Z6C+5ixaxEwMusr7mw@mail.gmail.com>
References: <CALg+e6z0=xhmohAJHEoapsP4oGgBzg1uyQgBe0NDJSeRrEa9RA@mail.gmail.com>
	<CALg+e6zRTn-=znDxQMhsTEojXCg+T8MhYcPHFbYSrRJirvq_vw@mail.gmail.com>
	<CAKV-jFyvQkfpMqGXihfe9ENJscQXWYs0Z6C+5ixaxEwMusr7mw@mail.gmail.com>
Message-ID: <1E02AA6015C24BEF955E114F8054FCE3@gmail.com>

FYI. I use R15B01 and never find a problem.  

-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Saturday, August 25, 2012 at 8:25 AM, Brian Foster wrote:

> It might be worth filing a JIRA so someone can work on this.
> 
> -BDF
> 
> On Fri, Aug 24, 2012 at 6:51 AM, Eranga Udesh <eranga.erl at gmail.com (mailto:eranga.erl at gmail.com)> wrote:
> > To answer my own question, the issue was with the Erlang version. It seels Mod Erlang Event is not compatible with R15B01 version. Once I downgrade the module compiling Erlang version to R14B03, it works fine.
> > 
> > Hope it helps others in the list.
> > 
> > - Eranga
> > 
> > 
> > 
> > 
> > On Thu, Aug 23, 2012 at 5:32 PM, Eranga Udesh <eranga.erl at gmail.com (mailto:eranga.erl at gmail.com)> wrote:
> > > Hi,
> > > 
> > > I'm trying mod_erlang_event module in Freeswitch. However when a call comes to extension 111, the Erlang node receives RPC call correctly (my config as below) but it seems Freeswitch module does not receive the returned {Ref, NewPid} tuple of the Erlang function. I use Erlang R15B01 version. 
> > > 
> > > I also tried with Registered process configuration as in the Mod Erlang Event wiki, but there also it timeout. I wonder if it is an incompatibility with Erlang 15B? 
> > > 
> > > Any ideas what could be wrong?
> > > 
> > > Config:
> > > > <extension name="to_erlang" continue="true"> 
> > > >    <condition field="destination_number" expression="^111$" >
> > > >        <action application="erlang" data="fsivr:new_call ivr at linux" />
> > > >    </condition>
> > > > </extension>
> > > > 
> > > > 
> > > Output in freeswitch debug:
> > > > 2012-08-23 17:06:34.057997 [DEBUG] mod_erlang_event.c:1519 enter erlang_outbound_function fsivr:new_call ivr at linux
> > > > 2012-08-23 17:06:34.057997 [DEBUG] mod_erlang_event.c:1525 Creating new listener for session
> > > > 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1527 Launching new listener
> > > > 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1537 Creating new spawned session for listener
> > > > 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1423 rpc call: fsivr:new_call(Ref)
> > > > 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1008 Connection Open
> > > > 2012-08-23 17:06:39.177997 [WARNING] mod_erlang_event.c:1436 Timed out when waiting for outbound pid 12.0.0 at freeswitch@linux 404fc3c7-45d1-4194-ad37-1e118fdee8d0
> > > > 2012-08-23 17:06:39.177997 [DEBUG] switch_channel.c:2919 (sofia/internal/5000 at linux4) Callstate Change RINGING -> HANGUP
> > > > 
> > > > 
> > > Thanks in advance.
> > > - Eranga
> > > 
> > > 
> > 
> > 
> > 
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > 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
> > 
> 
> 
> 
> -- 
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com (mailto:bdfoster at endigotech.com)
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
> 
> This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> 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
> 
> 


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From dujinfang at gmail.com  Sun Aug 26 02:36:05 2012
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 26 Aug 2012 06:36:05 +0800
Subject: [Freeswitch-users] FS 1.2.0 fails to build on OS X 10.8
In-Reply-To: <1C56216DA9394B4084D79C2B3E4F2F01@gmail.com>
References: <1C56216DA9394B4084D79C2B3E4F2F01@gmail.com>
Message-ID: <7F5F43FE9F164E2AB61B20B092C2AA55@gmail.com>

Yes, it has some issues but also workarounds on jira

http://www.google.com.hk/search?q=freeswitch+mountain+lion&rlz=1C1CHFA_enCN493CN493&sugexp=chrome,mod=14&sourceid=chrome&ie=UTF-8  

--  
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Friday, August 24, 2012 at 11:42 PM, urealfrank wrote:

>  
> Hi,
>  
> Anyone had success building FS 1.2.0 on Mountain Lion?
>  
> I did configure it this way:
>  
> $ git pull ; make clean ; ./bootstrap.sh (http://bootstrap.sh) ; ./configure --prefix=/usr/local/fs120 --disable-dependency-tracking --with-odbc=/usr/local/Cellar/unixodbc/2.3.1 --with-openssl ; make
>  
> ?
> Making all in ipt
> Making all in sdp
> Making all in url
> Making all in msg
> Making all in sip
> Making all in http
> Making all in soa
> Making all in tport
> LTCOMPILE tport_tls.lo
> cc1: warnings being treated as errors
> tport_tls.c: In function 'tls_init_once':
> tport_tls.c:98: warning: 'SSL_library_init' is deprecated (declared at /usr/include/openssl/ssl.h:1553)
> tport_tls.c:99: warning: 'SSL_load_error_strings' is deprecated (declared at /usr/include/openssl/ssl.h:1416)
> tport_tls.c:100: warning: 'SSL_get_ex_new_index' is deprecated (declared at /usr/include/openssl/ssl.h:1589)
> tport_tls.c: In function 'tls_log_errors':
> tport_tls.c:144: warning: 'ERR_get_error' is deprecated (declared at /usr/include/openssl/err.h:266)
> tport_tls.c:151: warning: 'ERR_get_error' is deprecated (declared at /usr/include/openssl/err.h:266)
> tport_tls.c:153: warning: 'ERR_lib_error_string' is deprecated (declared at /usr/include/openssl/err.h:281)
> tport_tls.c:154: warning: 'ERR_func_error_string' is deprecated (declared at /usr/include/openssl/err.h:282)
> tport_tls.c:155: warning: 'ERR_reason_error_string' is deprecated (declared at /usr/include/openssl/err.h:283)
> tport_tls.c: In function 'tls_verify_cb':
> tport_tls.c:220: warning: 'X509_STORE_CTX_get_current_cert' is deprecated (declared at /usr/include/openssl/x509_vfy.h:454)
> tport_tls.c:221: warning: 'X509_STORE_CTX_get_error_depth' is deprecated (declared at /usr/include/openssl/x509_vfy.h:453)
> tport_tls.c:222: warning: 'X509_STORE_CTX_get_error' is deprecated (declared at /usr/include/openssl/x509_vfy.h:451)
> tport_tls.c:223: warning: 'SSL_get_ex_data_X509_STORE_CTX_idx' is deprecated (declared at /usr/include/openssl/ssl.h:1601)
> tport_tls.c:224: warning: 'X509_STORE_CTX_get_ex_data' is deprecated (declared at /usr/include/openssl/x509_vfy.h:450)
> tport_tls.c:225: warning: 'SSL_get_ex_data' is deprecated (declared at /usr/include/openssl/ssl.h:1587)
> tport_tls.c:237: warning: 'X509_STORE_CTX_set_error' is deprecated (declared at /usr/include/openssl/x509_vfy.h:452)
> tport_tls.c:239: warning: 'X509_STORE_CTX_set_error' is deprecated (declared at /usr/include/openssl/x509_vfy.h:452)
> tport_tls.c:246: warning: 'X509_STORE_CTX_set_error' is deprecated (declared at /usr/include/openssl/x509_vfy.h:452)
> tport_tls.c:254: warning: 'X509_NAME_oneline' is deprecated (declared at /usr/include/openssl/x509.h:984)
> tport_tls.c:254: warning: 'X509_get_issuer_name' is deprecated (declared at /usr/include/openssl/x509.h:1011)
> tport_tls.c:256: warning: 'X509_NAME_oneline' is deprecated (declared at /usr/include/openssl/x509.h:984)
> tport_tls.c:256: warning: 'X509_get_subject_name' is deprecated (declared at /usr/include/openssl/x509.h:1013)
> tport_tls.c:258: warning: 'X509_verify_cert_error_string' is deprecated (declared at /usr/include/openssl/x509.h:752)
> tport_tls.c: In function 'tls_init_context':
> tport_tls.c:278: warning: 'RAND_load_file' is deprecated (declared at /usr/include/openssl/rand.h:108)
> tport_tls.c:301: warning: 'TLSv1_method' is deprecated (declared at /usr/include/openssl/ssl.h:1519)
> tport_tls.c:303: warning: 'SSLv23_method' is deprecated (declared at /usr/include/openssl/ssl.h:1515)
> tport_tls.c:305: warning: 'SSL_CTX_new' is deprecated (declared at /usr/include/openssl/ssl.h:1346)
> tport_tls.c:314: warning: 'SSL_CTX_set_timeout' is deprecated (declared at /usr/include/openssl/ssl.h:1348)
> tport_tls.c:318: warning: 'SSL_CTX_set_default_passwd_cb' is deprecated (declared at /usr/include/openssl/ssl.h:1472)
> tport_tls.c:319: warning: 'SSL_CTX_set_default_passwd_cb_userdata' is deprecated (declared at /usr/include/openssl/ssl.h:1473)
> tport_tls.c:322: warning: 'SSL_CTX_use_certificate_file' is deprecated (declared at /usr/include/openssl/ssl.h:1402)
> tport_tls.c:336: warning: 'SSL_CTX_use_PrivateKey_file' is deprecated (declared at /usr/include/openssl/ssl.h:1401)
> tport_tls.c:350: warning: 'SSL_CTX_check_private_key' is deprecated (declared at /usr/include/openssl/ssl.h:1475)
> tport_tls.c:361: warning: 'SSL_CTX_load_verify_locations' is deprecated (declared at /usr/include/openssl/ssl.h:1572)
> tport_tls.c:384: warning: 'SSL_CTX_set_verify_depth' is deprecated (declared at /usr/include/openssl/ssl.h:1460)
> tport_tls.c:385: warning: 'SSL_CTX_set_verify' is deprecated (declared at /usr/include/openssl/ssl.h:1459)
> tport_tls.c:387: warning: 'SSL_CTX_set_cipher_list' is deprecated (declared at /usr/include/openssl/ssl.h:1345)
> tport_tls.c: In function 'tls_free':
> tport_tls.c:403: warning: 'SSL_shutdown' is deprecated (declared at /usr/include/openssl/ssl.h:1532)
> tport_tls.c:406: warning: 'SSL_CTX_free' is deprecated (declared at /usr/include/openssl/ssl.h:1347)
> tport_tls.c:409: warning: 'BIO_free' is deprecated (declared at /usr/include/openssl/bio.h:583)
> tport_tls.c: In function 'tls_get_socket':
> tport_tls.c:419: warning: 'BIO_ctrl' is deprecated (declared at /usr/include/openssl/bio.h:590)
> tport_tls.c: In function 'tls_init_master':
> tport_tls.c:446: warning: 'RAND_pseudo_bytes' is deprecated (declared at /usr/include/openssl/rand.h:105)
> tport_tls.c:448: warning: 'SSL_CTX_set_session_id_context' is deprecated (declared at /usr/include/openssl/ssl.h:1479)
> tport_tls.c:453: warning: 'SSL_CTX_set_client_CA_list' is deprecated (declared at /usr/include/openssl/ssl.h:1542)
> tport_tls.c:454: warning: 'SSL_load_client_CA_file' is deprecated (declared at /usr/include/openssl/ssl.h:1404)
> tport_tls.c: In function 'tls_init_secondary':
> tport_tls.c:495: warning: 'BIO_new_socket' is deprecated (declared at /usr/include/openssl/bio.h:675)
> tport_tls.c:496: warning: 'SSL_new' is deprecated (declared at /usr/include/openssl/ssl.h:1481)
> tport_tls.c:505: warning: 'SSL_set_bio' is deprecated (declared at /usr/include/openssl/ssl.h:1375)
> tport_tls.c:506: warning: 'SSL_ctrl' is deprecated (declared at /usr/include/openssl/ssl.h:1496)
> tport_tls.c:507: warning: 'SSL_set_ex_data' is deprecated (declared at /usr/include/openssl/ssl.h:1586)
> tport_tls.c: In function 'tls_post_connection_check':
> tport_tls.c:523: warning: 'SSL_get_peer_certificate' is deprecated (declared at /usr/include/openssl/ssl.h:1450)
> tport_tls.c:539: warning: 'X509_get_ext_count' is deprecated (declared at /usr/include/openssl/x509.h:1144)
> tport_tls.c:554: warning: 'X509_get_ext' is deprecated (declared at /usr/include/openssl/x509.h:1148)
> tport_tls.c:555: warning: 'OBJ_nid2sn' is deprecated (declared at /usr/include/openssl/objects.h:1008)
> tport_tls.c:555: warning: 'OBJ_obj2nid' is deprecated (declared at /usr/include/openssl/objects.h:1009)
> tport_tls.c:555: warning: 'X509_EXTENSION_get_object' is deprecated (declared at /usr/include/openssl/x509.h:1185)
> tport_tls.c:560: warning: 'X509V3_EXT_get' is deprecated (declared at /usr/include/openssl/x509v3.h:581)
> tport_tls.c:561: warning: 'X509V3_EXT_d2i' is deprecated (declared at /usr/include/openssl/x509v3.h:585)
> tport_tls.c:564: warning: 'sk_num' is deprecated (declared at /usr/include/openssl/stack.h:81)
> tport_tls.c:565: warning: 'sk_value' is deprecated (declared at /usr/include/openssl/stack.h:82)
> tport_tls.c:579: warning: 'X509_get_subject_name' is deprecated (declared at /usr/include/openssl/x509.h:1013)
> tport_tls.c:582: warning: 'X509_NAME_get_text_by_NID' is deprecated (declared at /usr/include/openssl/x509.h:1099)
> tport_tls.c:599: warning: 'SSL_get_verify_result' is deprecated (declared at /usr/include/openssl/ssl.h:1584)
> tport_tls.c: In function 'tls_error':
> tport_tls.c:667: warning: 'SSL_get_error' is deprecated (declared at /usr/include/openssl/ssl.h:1501)
> tport_tls.c:682: warning: 'SSL_get_shutdown' is deprecated (declared at /usr/include/openssl/ssl.h:1568)
> tport_tls.c: In function 'tls_read':
> tport_tls.c:725: warning: 'SSL_read' is deprecated (declared at /usr/include/openssl/ssl.h:1493)
> tport_tls.c: In function 'tls_pending':
> tport_tls.c:741: warning: 'SSL_pending' is deprecated (declared at /usr/include/openssl/ssl.h:1368)
> tport_tls.c: In function 'tls_write':
> tport_tls.c:809: warning: 'SSL_write' is deprecated (declared at /usr/include/openssl/ssl.h:1495)
> tport_tls.c: In function 'tls_connect':
> tport_tls.c:901: warning: 'SSL_accept' is deprecated (declared at /usr/include/openssl/ssl.h:1491)
> tport_tls.c:901: warning: 'SSL_connect' is deprecated (declared at /usr/include/openssl/ssl.h:1492)
> tport_tls.c:902: warning: 'SSL_get_error' is deprecated (declared at /usr/include/openssl/ssl.h:1501)
> tport_tls.c:956: warning: 'ERR_error_string_n' is deprecated (declared at /usr/include/openssl/err.h:280)
> make[9]: *** [tport_tls.lo] Error 1
> make[8]: *** [all] Error 2
> Making all in nta
> Making all in nth
> Making all in nea
> Making all in iptsec
> Making all in nua
> make[8]: *** No rule to make target `tport/libtport.la (http://libtport.la)', needed by `libsofia-sip-ua.la (http://libsofia-sip-ua.la)'.  Stop.
> make[7]: *** [all-recursive] Error 1
> Making all in packages
> make[6]: *** [all-recursive] Error 1
> make[5]: *** [all] Error 2
> make[4]: *** [/Users/include/Documents/dev/Projects/WIMM/freeswitch_src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la (http://libsofia-sip-ua.la)] Error 2
> make[3]: *** [mod_sofia-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
>  
> $
>  
> Any magic potion? :)
>  
> Cheers
> Frank
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> 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
>  
>  


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From Nabble_01394 at slickdeals.endjunk.com  Sun Aug 26 04:44:36 2012
From: Nabble_01394 at slickdeals.endjunk.com (mazilo)
Date: Sat, 25 Aug 2012 17:44:36 -0700 (PDT)
Subject: [Freeswitch-users] Problem with linking freeswitch with GUI
In-Reply-To: <262109171-1345919753-cardhu_decombobulator_blackberry.rim.net-1468029409-@b13.c16.bise7.blackberry>
References: <CAHibNNek8QZ-J+GzuyxNHhnoJY5XFScW2Kf0q_ES5QmT4YNQBQ@mail.gmail.com>
	<1345902823870-7582206.post@n2.nabble.com>
	<262109171-1345919753-cardhu_decombobulator_blackberry.rim.net-1468029409-@b13.c16.bise7.blackberry>
Message-ID: <1345941876850-7582216.post@n2.nabble.com>


Ikenna Onyeagba wrote
> 
> Thanks for ur reply, I've got freeswitch already installed, just to be
> sure, am I goin to install the GUI as a seperate installation from d
> freeswitch?
Great. Your next step will probably be getting a copy of FusionPBX source
code to install on your FS system (I used source codes from the SVN trunk).
Then, you will need to configure the web server running on your system that
hosts the FusionPBX + FS. That's all that to it. If your configuration is
correct and your system is connected to your private local LAN, your web
server shall launch your FusionPBX  for anyone to access from any web
browser on any computers connected to the same local private LAN.

The version of FusionPBX I had tested on my Seagate DockStar (a few months
ago) was rather a v2.x. Unlike Blue.Box, FusionPBX 2.x is very responsive on
my Seagate DockStar.



-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-linking-freeswitch-with-GUI-tp7582162p7582216.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From b2m at a-cti.com  Sun Aug 26 05:34:09 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Sun, 26 Aug 2012 07:04:09 +0530
Subject: [Freeswitch-users] Recording Volume
In-Reply-To: <1345906479516-7582207.post@n2.nabble.com>
References: <CAP2fbbcfO8Oo_SQ4AOmUb2W3L+oTZWtpjffAustdAGNeB_dQBg@mail.gmail.com>
	<1345906479516-7582207.post@n2.nabble.com>
Message-ID: <CAP2fbbdYnqB3Rtc4J+7x_8-2NjqbR3UveDJ6ac0Fv4YRwCoA-g@mail.gmail.com>

Call volume is fine, recording volume need to be tuned.

Thanks,
Bala

On Saturday, August 25, 2012, Jeff Lenk wrote:

> Call volume should be adjusted at the source not at the destination. You
> should investigate why fs is receiving audio that is not loud enough.
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Recording-Volume-tp7582204p7582207.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.olsson at visionutveckling.se  Sun Aug 26 12:13:34 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 26 Aug 2012 08:13:34 +0000
Subject: [Freeswitch-users] Recording Volume
In-Reply-To: <CAP2fbbdYnqB3Rtc4J+7x_8-2NjqbR3UveDJ6ac0Fv4YRwCoA-g@mail.gmail.com>
References: <CAP2fbbcfO8Oo_SQ4AOmUb2W3L+oTZWtpjffAustdAGNeB_dQBg@mail.gmail.com>
	<1345906479516-7582207.post@n2.nabble.com>,
	<CAP2fbbdYnqB3Rtc4J+7x_8-2NjqbR3UveDJ6ac0Fv4YRwCoA-g@mail.gmail.com>
Message-ID: <1FFF97C269757C458224B7C895F35F1514EAB7@cantor.std.visionutv.se>

FS records exactly as-is (no volume tuning), so if call volume is ok, the recorded volume is exactly the same. If this is not true, you must have something really strange going on in your endpoints.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Balamurugan Mahendran [b2m at a-cti.com]
Skickat: den 26 augusti 2012 03:34
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Recording Volume

Call volume is fine, recording volume need to be tuned.

Thanks,
Bala

On Saturday, August 25, 2012, Jeff Lenk wrote:
Call volume should be adjusted at the source not at the destination. You
should investigate why fs is receiving audio that is not loud enough.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Recording-Volume-tp7582204p7582207.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<UrlBlockedError.aspx>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<UrlBlockedError.aspx>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From urealfrank at gmail.com  Sun Aug 26 17:24:16 2012
From: urealfrank at gmail.com (urealfrank)
Date: Sun, 26 Aug 2012 14:24:16 +0100
Subject: [Freeswitch-users] FS 1.2.0 fails to build on OS X 10.8
In-Reply-To: <7F5F43FE9F164E2AB61B20B092C2AA55@gmail.com>
References: <1C56216DA9394B4084D79C2B3E4F2F01@gmail.com>
	<7F5F43FE9F164E2AB61B20B092C2AA55@gmail.com>
Message-ID: <0CEDB91EA08940A49E936CABF39C43EF@gmail.com>

Thanks for your help :)

I did that search before posting my question here... The thing is, I've installed unixodbc, i've used it after comment both spidermonkey and speex modules, the result is the same...  


Cheers
Frank


On Saturday, August 25, 2012 at 11:36 PM, Seven Du wrote:

> Yes, it has some issues but also workarounds on jira
>  
> http://www.google.com.hk/search?q=freeswitch+mountain+lion&rlz=1C1CHFA_enCN493CN493&sugexp=chrome,mod=14&sourceid=chrome&ie=UTF-8  
>  
> --  
> Seven Du
> Sent with Sparrow (http://www.sparrowmailapp.com/?sig)
>  
>  
> On Friday, August 24, 2012 at 11:42 PM, urealfrank wrote:
>  
> >  
> > Hi,
> >  
> > Anyone had success building FS 1.2.0 on Mountain Lion?
> >  
> > I did configure it this way:
> >  
> > $ git pull ; make clean ; ./bootstrap.sh (http://bootstrap.sh) ; ./configure --prefix=/usr/local/fs120 --disable-dependency-tracking --with-odbc=/usr/local/Cellar/unixodbc/2.3.1 --with-openssl ; make
> >  
> > ?
> > Making all in ipt
> > Making all in sdp
> > Making all in url
> > Making all in msg
> > Making all in sip
> > Making all in http
> > Making all in soa
> > Making all in tport
> > LTCOMPILE tport_tls.lo
> > cc1: warnings being treated as errors
> > tport_tls.c: In function 'tls_init_once':
> > tport_tls.c:98: warning: 'SSL_library_init' is deprecated (declared at /usr/include/openssl/ssl.h:1553)
> > tport_tls.c:99: warning: 'SSL_load_error_strings' is deprecated (declared at /usr/include/openssl/ssl.h:1416)
> > tport_tls.c:100: warning: 'SSL_get_ex_new_index' is deprecated (declared at /usr/include/openssl/ssl.h:1589)
> > tport_tls.c: In function 'tls_log_errors':
> > tport_tls.c:144: warning: 'ERR_get_error' is deprecated (declared at /usr/include/openssl/err.h:266)
> > tport_tls.c:151: warning: 'ERR_get_error' is deprecated (declared at /usr/include/openssl/err.h:266)
> > tport_tls.c:153: warning: 'ERR_lib_error_string' is deprecated (declared at /usr/include/openssl/err.h:281)
> > tport_tls.c:154: warning: 'ERR_func_error_string' is deprecated (declared at /usr/include/openssl/err.h:282)
> > tport_tls.c:155: warning: 'ERR_reason_error_string' is deprecated (declared at /usr/include/openssl/err.h:283)
> > tport_tls.c: In function 'tls_verify_cb':
> > tport_tls.c:220: warning: 'X509_STORE_CTX_get_current_cert' is deprecated (declared at /usr/include/openssl/x509_vfy.h:454)
> > tport_tls.c:221: warning: 'X509_STORE_CTX_get_error_depth' is deprecated (declared at /usr/include/openssl/x509_vfy.h:453)
> > tport_tls.c:222: warning: 'X509_STORE_CTX_get_error' is deprecated (declared at /usr/include/openssl/x509_vfy.h:451)
> > tport_tls.c:223: warning: 'SSL_get_ex_data_X509_STORE_CTX_idx' is deprecated (declared at /usr/include/openssl/ssl.h:1601)
> > tport_tls.c:224: warning: 'X509_STORE_CTX_get_ex_data' is deprecated (declared at /usr/include/openssl/x509_vfy.h:450)
> > tport_tls.c:225: warning: 'SSL_get_ex_data' is deprecated (declared at /usr/include/openssl/ssl.h:1587)
> > tport_tls.c:237: warning: 'X509_STORE_CTX_set_error' is deprecated (declared at /usr/include/openssl/x509_vfy.h:452)
> > tport_tls.c:239: warning: 'X509_STORE_CTX_set_error' is deprecated (declared at /usr/include/openssl/x509_vfy.h:452)
> > tport_tls.c:246: warning: 'X509_STORE_CTX_set_error' is deprecated (declared at /usr/include/openssl/x509_vfy.h:452)
> > tport_tls.c:254: warning: 'X509_NAME_oneline' is deprecated (declared at /usr/include/openssl/x509.h:984)
> > tport_tls.c:254: warning: 'X509_get_issuer_name' is deprecated (declared at /usr/include/openssl/x509.h:1011)
> > tport_tls.c:256: warning: 'X509_NAME_oneline' is deprecated (declared at /usr/include/openssl/x509.h:984)
> > tport_tls.c:256: warning: 'X509_get_subject_name' is deprecated (declared at /usr/include/openssl/x509.h:1013)
> > tport_tls.c:258: warning: 'X509_verify_cert_error_string' is deprecated (declared at /usr/include/openssl/x509.h:752)
> > tport_tls.c: In function 'tls_init_context':
> > tport_tls.c:278: warning: 'RAND_load_file' is deprecated (declared at /usr/include/openssl/rand.h:108)
> > tport_tls.c:301: warning: 'TLSv1_method' is deprecated (declared at /usr/include/openssl/ssl.h:1519)
> > tport_tls.c:303: warning: 'SSLv23_method' is deprecated (declared at /usr/include/openssl/ssl.h:1515)
> > tport_tls.c:305: warning: 'SSL_CTX_new' is deprecated (declared at /usr/include/openssl/ssl.h:1346)
> > tport_tls.c:314: warning: 'SSL_CTX_set_timeout' is deprecated (declared at /usr/include/openssl/ssl.h:1348)
> > tport_tls.c:318: warning: 'SSL_CTX_set_default_passwd_cb' is deprecated (declared at /usr/include/openssl/ssl.h:1472)
> > tport_tls.c:319: warning: 'SSL_CTX_set_default_passwd_cb_userdata' is deprecated (declared at /usr/include/openssl/ssl.h:1473)
> > tport_tls.c:322: warning: 'SSL_CTX_use_certificate_file' is deprecated (declared at /usr/include/openssl/ssl.h:1402)
> > tport_tls.c:336: warning: 'SSL_CTX_use_PrivateKey_file' is deprecated (declared at /usr/include/openssl/ssl.h:1401)
> > tport_tls.c:350: warning: 'SSL_CTX_check_private_key' is deprecated (declared at /usr/include/openssl/ssl.h:1475)
> > tport_tls.c:361: warning: 'SSL_CTX_load_verify_locations' is deprecated (declared at /usr/include/openssl/ssl.h:1572)
> > tport_tls.c:384: warning: 'SSL_CTX_set_verify_depth' is deprecated (declared at /usr/include/openssl/ssl.h:1460)
> > tport_tls.c:385: warning: 'SSL_CTX_set_verify' is deprecated (declared at /usr/include/openssl/ssl.h:1459)
> > tport_tls.c:387: warning: 'SSL_CTX_set_cipher_list' is deprecated (declared at /usr/include/openssl/ssl.h:1345)
> > tport_tls.c: In function 'tls_free':
> > tport_tls.c:403: warning: 'SSL_shutdown' is deprecated (declared at /usr/include/openssl/ssl.h:1532)
> > tport_tls.c:406: warning: 'SSL_CTX_free' is deprecated (declared at /usr/include/openssl/ssl.h:1347)
> > tport_tls.c:409: warning: 'BIO_free' is deprecated (declared at /usr/include/openssl/bio.h:583)
> > tport_tls.c: In function 'tls_get_socket':
> > tport_tls.c:419: warning: 'BIO_ctrl' is deprecated (declared at /usr/include/openssl/bio.h:590)
> > tport_tls.c: In function 'tls_init_master':
> > tport_tls.c:446: warning: 'RAND_pseudo_bytes' is deprecated (declared at /usr/include/openssl/rand.h:105)
> > tport_tls.c:448: warning: 'SSL_CTX_set_session_id_context' is deprecated (declared at /usr/include/openssl/ssl.h:1479)
> > tport_tls.c:453: warning: 'SSL_CTX_set_client_CA_list' is deprecated (declared at /usr/include/openssl/ssl.h:1542)
> > tport_tls.c:454: warning: 'SSL_load_client_CA_file' is deprecated (declared at /usr/include/openssl/ssl.h:1404)
> > tport_tls.c: In function 'tls_init_secondary':
> > tport_tls.c:495: warning: 'BIO_new_socket' is deprecated (declared at /usr/include/openssl/bio.h:675)
> > tport_tls.c:496: warning: 'SSL_new' is deprecated (declared at /usr/include/openssl/ssl.h:1481)
> > tport_tls.c:505: warning: 'SSL_set_bio' is deprecated (declared at /usr/include/openssl/ssl.h:1375)
> > tport_tls.c:506: warning: 'SSL_ctrl' is deprecated (declared at /usr/include/openssl/ssl.h:1496)
> > tport_tls.c:507: warning: 'SSL_set_ex_data' is deprecated (declared at /usr/include/openssl/ssl.h:1586)
> > tport_tls.c: In function 'tls_post_connection_check':
> > tport_tls.c:523: warning: 'SSL_get_peer_certificate' is deprecated (declared at /usr/include/openssl/ssl.h:1450)
> > tport_tls.c:539: warning: 'X509_get_ext_count' is deprecated (declared at /usr/include/openssl/x509.h:1144)
> > tport_tls.c:554: warning: 'X509_get_ext' is deprecated (declared at /usr/include/openssl/x509.h:1148)
> > tport_tls.c:555: warning: 'OBJ_nid2sn' is deprecated (declared at /usr/include/openssl/objects.h:1008)
> > tport_tls.c:555: warning: 'OBJ_obj2nid' is deprecated (declared at /usr/include/openssl/objects.h:1009)
> > tport_tls.c:555: warning: 'X509_EXTENSION_get_object' is deprecated (declared at /usr/include/openssl/x509.h:1185)
> > tport_tls.c:560: warning: 'X509V3_EXT_get' is deprecated (declared at /usr/include/openssl/x509v3.h:581)
> > tport_tls.c:561: warning: 'X509V3_EXT_d2i' is deprecated (declared at /usr/include/openssl/x509v3.h:585)
> > tport_tls.c:564: warning: 'sk_num' is deprecated (declared at /usr/include/openssl/stack.h:81)
> > tport_tls.c:565: warning: 'sk_value' is deprecated (declared at /usr/include/openssl/stack.h:82)
> > tport_tls.c:579: warning: 'X509_get_subject_name' is deprecated (declared at /usr/include/openssl/x509.h:1013)
> > tport_tls.c:582: warning: 'X509_NAME_get_text_by_NID' is deprecated (declared at /usr/include/openssl/x509.h:1099)
> > tport_tls.c:599: warning: 'SSL_get_verify_result' is deprecated (declared at /usr/include/openssl/ssl.h:1584)
> > tport_tls.c: In function 'tls_error':
> > tport_tls.c:667: warning: 'SSL_get_error' is deprecated (declared at /usr/include/openssl/ssl.h:1501)
> > tport_tls.c:682: warning: 'SSL_get_shutdown' is deprecated (declared at /usr/include/openssl/ssl.h:1568)
> > tport_tls.c: In function 'tls_read':
> > tport_tls.c:725: warning: 'SSL_read' is deprecated (declared at /usr/include/openssl/ssl.h:1493)
> > tport_tls.c: In function 'tls_pending':
> > tport_tls.c:741: warning: 'SSL_pending' is deprecated (declared at /usr/include/openssl/ssl.h:1368)
> > tport_tls.c: In function 'tls_write':
> > tport_tls.c:809: warning: 'SSL_write' is deprecated (declared at /usr/include/openssl/ssl.h:1495)
> > tport_tls.c: In function 'tls_connect':
> > tport_tls.c:901: warning: 'SSL_accept' is deprecated (declared at /usr/include/openssl/ssl.h:1491)
> > tport_tls.c:901: warning: 'SSL_connect' is deprecated (declared at /usr/include/openssl/ssl.h:1492)
> > tport_tls.c:902: warning: 'SSL_get_error' is deprecated (declared at /usr/include/openssl/ssl.h:1501)
> > tport_tls.c:956: warning: 'ERR_error_string_n' is deprecated (declared at /usr/include/openssl/err.h:280)
> > make[9]: *** [tport_tls.lo] Error 1
> > make[8]: *** [all] Error 2
> > Making all in nta
> > Making all in nth
> > Making all in nea
> > Making all in iptsec
> > Making all in nua
> > make[8]: *** No rule to make target `tport/libtport.la (http://libtport.la)', needed by `libsofia-sip-ua.la (http://libsofia-sip-ua.la)'.  Stop.
> > make[7]: *** [all-recursive] Error 1
> > Making all in packages
> > make[6]: *** [all-recursive] Error 1
> > make[5]: *** [all] Error 2
> > make[4]: *** [/Users/include/Documents/dev/Projects/WIMM/freeswitch_src/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la (http://libsofia-sip-ua.la)] Error 2
> > make[3]: *** [mod_sofia-all] Error 1
> > make[2]: *** [all-recursive] Error 1
> > make[1]: *** [all-recursive] Error 1
> > make: *** [all] Error 2
> >  
> > $
> >  
> > Any magic potion? :)
> >  
> > Cheers
> > Frank
> >  
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > http://www.freeswitchsolutions.com
> >  
> > 
> > 
> >  
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > 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
> >  
> >  
> >  
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> 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
>  
>  
>  


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From bdfoster at endigotech.com  Mon Aug 27 03:41:22 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 26 Aug 2012 19:41:22 -0400
Subject: [Freeswitch-users] Polycom Registration Failing
Message-ID: <CAKV-jFwqp9R=j2bEv7fQnL4ticFTXBeJEvjmyCyDoL3wJXUt=w@mail.gmail.com>

I have a Polycom IP 335 and a Linksys SPA921 here at my office (on the same
desk, connected to the same hosted PBX; don't ask). The Linksys is not
having issues registering. The Polycom, however, gets a 401 Unauthorized
every time it tries to register. Both phones are behind the same NAT, both
are registering to the same PBX, and they are registering to the same
extension.

SIP messages here: http://pastebin.freeswitch.org/19765
<http://pastebin.freeswitch.org/19765>Internal Sofia Profile here:
http://pastebin.freeswitch.org/19766

Any hints? I've reset both phones, updated to latest in v1.2.stable, etc.
Still no joy.

-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
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From haloha201 at gmail.com  Mon Aug 27 07:20:31 2012
From: haloha201 at gmail.com (haloha)
Date: Mon, 27 Aug 2012 10:20:31 +0700
Subject: [Freeswitch-users] need help on gateway
Message-ID: <CAMBhhBK6sgPckoxa8JZ1Tpe-60+K-imaWXfc+7i0jHWo=RfteQ@mail.gmail.com>

Hi all

for now, all the gateway's information is stored in xml file and its
path(conf/sip_profiles/external/example.xml):
<include>
	  <gateway name="example">
		  <param name="username" value="cluecon"/>
		  <param name="password" value="2007"/>
		  <param name="proxy" value="sip.abc.net"/>
		  <param name="expire-seconds" value="3600"/>
		  <param name="register" value="true"/>
		  <!--<param name="register" value="false"/>-->
		  <param name="retry-seconds" value="30"/>
		  <param name="ping" value="25"/>
	  </gateway>
</include>

is there a way to store the gateway's information in Mysql Databases,
like sip clients information in folder conf/directory/

Thank you
Ha`


From bdfoster at endigotech.com  Mon Aug 27 07:25:24 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 26 Aug 2012 23:25:24 -0400
Subject: [Freeswitch-users] need help on gateway
In-Reply-To: <CAMBhhBK6sgPckoxa8JZ1Tpe-60+K-imaWXfc+7i0jHWo=RfteQ@mail.gmail.com>
References: <CAMBhhBK6sgPckoxa8JZ1Tpe-60+K-imaWXfc+7i0jHWo=RfteQ@mail.gmail.com>
Message-ID: <CAKV-jFzcmu29-1LmvpU0MbqJER31jQzzJFy9X4AK6SDytSvLKg@mail.gmail.com>

Easiest way is to use xml curl, checkout mod_xml_curl. You can pull things
from mysql via your scripts on the web server.

-BDF

On Sun, Aug 26, 2012 at 11:20 PM, haloha <haloha201 at gmail.com> wrote:

> Hi all
>
> for now, all the gateway's information is stored in xml file and its
> path(conf/sip_profiles/external/example.xml):
> <include>
>           <gateway name="example">
>                   <param name="username" value="cluecon"/>
>                   <param name="password" value="2007"/>
>                   <param name="proxy" value="sip.abc.net"/>
>                   <param name="expire-seconds" value="3600"/>
>                   <param name="register" value="true"/>
>                   <!--<param name="register" value="false"/>-->
>                   <param name="retry-seconds" value="30"/>
>                   <param name="ping" value="25"/>
>           </gateway>
> </include>
>
> is there a way to store the gateway's information in Mysql Databases,
> like sip clients information in folder conf/directory/
>
> Thank you
> Ha`
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
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From haloha201 at gmail.com  Mon Aug 27 08:16:50 2012
From: haloha201 at gmail.com (haloha)
Date: Mon, 27 Aug 2012 11:16:50 +0700
Subject: [Freeswitch-users] need help on gateway
In-Reply-To: <CAKV-jFzcmu29-1LmvpU0MbqJER31jQzzJFy9X4AK6SDytSvLKg@mail.gmail.com>
References: <CAMBhhBK6sgPckoxa8JZ1Tpe-60+K-imaWXfc+7i0jHWo=RfteQ@mail.gmail.com>
	<CAKV-jFzcmu29-1LmvpU0MbqJER31jQzzJFy9X4AK6SDytSvLKg@mail.gmail.com>
Message-ID: <CAMBhhBLdow__4Dk_ymVsxp4PoQTYhQ1WHD4c3qC58p6OHqnUhw@mail.gmail.com>

hi Brian

could you please give me the example of the respond for gateway's info
from web server
there is no example for gateway's info respond,
i only found the respond for user's info:
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="domain1.awesomevoipdomain.faketld">
      <params>
        <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
      </params>
      <groups>
        <group name="default">
         <users>
          <user id="1004">
            <params>
              <param name="password" value="some_password"/>
            </params>
          </user>
         </users>
        </group>
      </groups>
    </domain>
  </section>
</document>

Thank you
Ha`

On 8/27/12, Brian Foster <bdfoster at endigotech.com> wrote:
> Easiest way is to use xml curl, checkout mod_xml_curl. You can pull things
> from mysql via your scripts on the web server.
>
> -BDF
>
> On Sun, Aug 26, 2012 at 11:20 PM, haloha <haloha201 at gmail.com> wrote:
>
>> Hi all
>>
>> for now, all the gateway's information is stored in xml file and its
>> path(conf/sip_profiles/external/example.xml):
>> <include>
>>           <gateway name="example">
>>                   <param name="username" value="cluecon"/>
>>                   <param name="password" value="2007"/>
>>                   <param name="proxy" value="sip.abc.net"/>
>>                   <param name="expire-seconds" value="3600"/>
>>                   <param name="register" value="true"/>
>>                   <!--<param name="register" value="false"/>-->
>>                   <param name="retry-seconds" value="30"/>
>>                   <param name="ping" value="25"/>
>>           </gateway>
>> </include>
>>
>> is there a way to store the gateway's information in Mysql Databases,
>> like sip clients information in folder conf/directory/
>>
>> Thank you
>> Ha`
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>


From haloha201 at gmail.com  Mon Aug 27 11:07:28 2012
From: haloha201 at gmail.com (haloha)
Date: Mon, 27 Aug 2012 14:07:28 +0700
Subject: [Freeswitch-users] need help on gateway
In-Reply-To: <CAMBhhBLdow__4Dk_ymVsxp4PoQTYhQ1WHD4c3qC58p6OHqnUhw@mail.gmail.com>
References: <CAMBhhBK6sgPckoxa8JZ1Tpe-60+K-imaWXfc+7i0jHWo=RfteQ@mail.gmail.com>
	<CAKV-jFzcmu29-1LmvpU0MbqJER31jQzzJFy9X4AK6SDytSvLKg@mail.gmail.com>
	<CAMBhhBLdow__4Dk_ymVsxp4PoQTYhQ1WHD4c3qC58p6OHqnUhw@mail.gmail.com>
Message-ID: <CAMBhhBLG82V1AngMvgu1fckk8tqLTqO_5yhqopiw5=E-bU7FbA@mail.gmail.com>

Hi

i did cli in freeswitch mode: sofia profile external rescan
and freeswitch sends to web server

hostname=localhost.localdomain
section=directory
tag_name=
key_name=
key_value=
Event-Name=REQUEST_PARAMS
Core-UUID=4144f203-0c8f-4050-b58b-a39564428f3a
FreeSWITCH-Hostname=localhost.localdomain
FreeSWITCH-Switchname=localhost.localdomain
FreeSWITCH-IPv4=172.30.41.148
FreeSWITCH-IPv6=%3A%3A1
Event-Date-Local=2012-08-27%2003%3A00%3A19
Event-Date-GMT=Mon,%2027%20Aug%202012%2007%3A00%3A19%20GMT
Event-Date-Timestamp=1346050819261656
Event-Calling-File=sofia.c
Event-Calling-Function=launch_sofia_worker_thread
Event-Calling-Line-Number=2006
Event-Sequence=431
purpose=gateways
profile=external



i try to send back

<?xml version="1.0" encoding="utf-8"?>
<document type="freeswitch/xml">
  <section name="directory">
   <include>
    <gateway name="example">
     <param name="username" value="cluecon"/>
     <param name="password" value="2007"/>
     <param name="proxy" value="sip.abc.net"/>
     <param name="expire-seconds" value="3600"/>
     <param name="register" value="false"/>
     <param name="retry-seconds" value="30"/>
     <param name="ping" value="25"/>
    </gateway>
   </include>
  </section>
</document>

but it does not work

what is the correct format form

Thank you
Ha`





On 8/27/12, haloha <haloha201 at gmail.com> wrote:
> hi Brian
>
> could you please give me the example of the respond for gateway's info
> from web server
> there is no example for gateway's info respond,
> i only found the respond for user's info:
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="domain1.awesomevoipdomain.faketld">
>       <params>
>         <param name="dial-string"
> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>       </params>
>       <groups>
>         <group name="default">
>          <users>
>           <user id="1004">
>             <params>
>               <param name="password" value="some_password"/>
>             </params>
>           </user>
>          </users>
>         </group>
>       </groups>
>     </domain>
>   </section>
> </document>
>
> Thank you
> Ha`
>
> On 8/27/12, Brian Foster <bdfoster at endigotech.com> wrote:
>> Easiest way is to use xml curl, checkout mod_xml_curl. You can pull
>> things
>> from mysql via your scripts on the web server.
>>
>> -BDF
>>
>> On Sun, Aug 26, 2012 at 11:20 PM, haloha <haloha201 at gmail.com> wrote:
>>
>>> Hi all
>>>
>>> for now, all the gateway's information is stored in xml file and its
>>> path(conf/sip_profiles/external/example.xml):
>>> <include>
>>>           <gateway name="example">
>>>                   <param name="username" value="cluecon"/>
>>>                   <param name="password" value="2007"/>
>>>                   <param name="proxy" value="sip.abc.net"/>
>>>                   <param name="expire-seconds" value="3600"/>
>>>                   <param name="register" value="true"/>
>>>                   <!--<param name="register" value="false"/>-->
>>>                   <param name="retry-seconds" value="30"/>
>>>                   <param name="ping" value="25"/>
>>>           </gateway>
>>> </include>
>>>
>>> is there a way to store the gateway's information in Mysql Databases,
>>> like sip clients information in folder conf/directory/
>>>
>>> Thank you
>>> Ha`
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>>
>>
>> --
>> Brian D. Foster
>> Endigo Computer LLC
>> Email: bdfoster at endigotech.com
>> Phone: 317-800-7876
>> Indianapolis, Indiana, USA
>>
>> This message contains confidential information and is intended for those
>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header.
>> If
>> you are not the intended recipient you are notified that disclosing,
>> copying, distributing or taking any action in reliance on the contents of
>> this information is strictly prohibited. E-mail transmission cannot be
>> guaranteed to be secure or error-free as information could be
>> intercepted,
>> corrupted, lost, destroyed, arrive late or incomplete, or contain
>> viruses.
>> The sender therefore does not accept liability for any errors or
>> omissions
>> in the contents of this message, which arise as a result of e-mail
>> transmission. If verification is required please request a hard-copy
>> version.
>>
>


From gerald.weber at besharp.at  Mon Aug 27 13:01:42 2012
From: gerald.weber at besharp.at (Gerald Weber)
Date: Mon, 27 Aug 2012 09:01:42 +0000
Subject: [Freeswitch-users] mod_callcenter Design Questions
Message-ID: <D1925DECD316894DAADB84076C1B7D827E28DC94@exchange.besharp.at>

Hi all,

some questions regarding design decisions in mod_callcenter,primary adressed to moc.

1) why did you put the queues in a hash table rather than in db tables ?

2) what do you recommend for the following scenario:

when an agent call ends, there should be an event for the wrap-up-time start and one for wrap-up-time end.
i implemented sending end of the event using switch_scheduler_add_task, so a new task is started with the given wrap-up-time
when the call ends. this works, but has 2 disadvantages:
- if the agent ends the wrapuptime by hand, the scheduler task keeps running.-
- i cannot extend the wrapuptime on the fly if needed (or is there a fs api call to extend the time ? )

so my next idea is to put an additional sql query into cc_agent_dispatch_thread_run and do the time calculations in the callback (also for onbreak)
do you think this is a better approach ?

3) did you consider setting the agentstate for outbound / inbound calls from / to the agent contact(phone) ?

thanks&regards

gw
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From itispip-qq at hotmail.com  Mon Aug 27 08:47:03 2012
From: itispip-qq at hotmail.com (Pip Live)
Date: Mon, 27 Aug 2012 12:47:03 +0800
Subject: [Freeswitch-users] How to know inbound call arrive from which
	gateway?
Message-ID: <BLU0-SMTP4659A4A0496E38F9F7E779D8CA20@phx.gbl>

Sorry this maybe a very simple question but I checked Wiki/Dialplan without
finding an answer.

How can I know inbound call arrive at which sofia gateway in dialplan? My
purpose is to add a prefix to all inbound call from a certain gateway. I
cannot use the "destination_number" built_in variable for this purpose, as
in my case DID bind to a specific gateway hardware is often changing. Is
that another varialbe to be used, eg. a gateway name etc?

/Alex
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From wingcomm at hotmail.com  Mon Aug 27 09:10:50 2012
From: wingcomm at hotmail.com (R W)
Date: Mon, 27 Aug 2012 01:10:50 -0400
Subject: [Freeswitch-users] Calls drop after ~1:48
Message-ID: <BAY156-W267FC7C8D62A763110D51FD4A20@phx.gbl>


We're seeing this same exact issue with a number of our phones (specifically CounterPath Bria for iOS and others). Calls placed disconnect after ~1:48 on some NATs, (specifically our wireless data carrier, AT&T 3G).

http://lists.freeswitch.org/pipermail/freeswitch-users/2012-March/081317.html
In freeswitch, the CLI shows:
















2012-08-27 00:41:59.471201 [DEBUG] switch_core_session.c:1429 Session 174 (sofia/internal/2000 at mydomain.com) Locked, Waiting on external entities
2012-08-27 00:41:59.471201 [NOTICE] switch_core_session.c:1447 Session 174 (sofia/internal/2000 at mydomain.com) Ended
2012-08-27 00:41:59.471201 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/2000 at mydomain.com [CS_DESTROY]
2012-08-27 00:41:59.471201 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/2000 at mydomain.com) Callstate Change HANGUP -> DOWN
2012-08-27 00:41:59.471201 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/2000 at mydomain.com) Running State Change CS_DESTROY
2012-08-27 00:41:59.471201 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/2000 at mydomain.com) State DESTROY
2012-08-27 00:41:59.471201 [DEBUG] mod_sofia.c:374 sofia/internal/2000 at mydomain.com SOFIA DESTROY
2012-08-27 00:41:59.471201 [DEBUG] switch_core_state_machine.c:86 sofia/internal/2000 at mydomain.com Standard DESTROY
2012-08-27 00:41:59.471201 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/2000 at mydomain.com) State DESTROY going to sleep
We fixed this by shutting of the "Passive Session Timer" in the "SIP Miscellaneous." However, I'm curious to know if anyone can explain why this happens on some NATs and not others? Is there s problem with disabling the "Passive Session Timer" on our clients?
-Rob

 		 	   		  
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From wingcomm at hotmail.com  Mon Aug 27 10:34:17 2012
From: wingcomm at hotmail.com (R W)
Date: Mon, 27 Aug 2012 02:34:17 -0400
Subject: [Freeswitch-users] Calls from SRTP Clients to non-SRTP clients
In-Reply-To: <CAN5esQJUoQ2RFgAih=Wd59V=dMuh3od1bobnw4jAA_ymM1btoQ@mail.gmail.com>
References: <BAY156-W3667570B62D531ABCA6E7FD4B70@phx.gbl>,
	<CAN5esQJUoQ2RFgAih=Wd59V=dMuh3od1bobnw4jAA_ymM1btoQ@mail.gmail.com>
Message-ID: <BAY156-W646521E6E88ED0988DB81DD4A20@phx.gbl>


Richard et al.,
Thank you for your insight. To answer your last question, I did refer to Bria for the iPhone and Bria 3 on Mac OS X since I assumed this issue was limited to non-SRTP devices and SRTP devices. However, further testing, however has shown what appears to be an incompatibility between Bria for the iPhone and Bria for Mac OS X.
Both devices can make/receive calls through FreeSWITCH to external providers and to services like voicemail and I can verify that SRTP is active between FreeSWITCH <-> Bria. I can even make calls to between Bria for iOS (iPhone/iPad) to other Bria for iOS devices with SRTP...
However with SRTP enabled/required on both Bria softphones (iOS/iPhone/iPad and Mac OS X) I get a "SERVICE_NOT_IMPLEMENTED" in FreeSWITCH when I call Bria (iOS/Mac OS X) to Bria (iOS/Mac OS X).
2012-08-27 02:16:33.501940 [NOTICE] sofia.c:6847 Hangup sofia/internal/sip:2003@<ip removed>:63615 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]...2012-08-27 02:16:33.501940 [DEBUG] switch_ivr_originate.c:3458 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]2012-08-27 02:16:33.501940 [NOTICE] switch_ivr_originate.c:2544 Cannot create outgoing channel of type [user] cause: [SERVICE_NOT_IMPLEMENTED]2012-08-27 02:16:33.501940 [DEBUG] switch_ivr_originate.c:3458 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]2012-08-27 02:16:33.501940 [INFO] mod_dptools.c:3027 Originate Failed.  Cause: SERVICE_NOT_IMPLEMENTED
I am going to do a little further testing, but any insight into what this could be would be appreciated. Has anyone else seen this?
Best,
-Rob
Date: Tue, 14 Aug 2012 18:41:03 +0100
From: rnbrady at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Calls from SRTP Clients to non-SRTP clients

> is there a way to force FreeSWITCH to establish an SRTP call to clients when the originating client does not support SRTP?
This should work by default, assuming you are setting sip_secure_media in the appropriate place.

FreeSWITCH should negotiate both channels (legs) independently. So if the A-end has no SRTP, that should not prevent FreeSWITCH from sending a INVITE to the B-end with SRTP specified (i.e. SAVP in the SDP with a crypto attribute).

I think "all or nothing" doesn't imply both ends of the call, it implies all calls or none of the calls calls. So an inbound or outbound call without SRTP will be rejected. Hope this makes sense.

However, in the default dialplan there is a condition that will cause FreeSWITCH to implement such a policy. It is commented out by default:
      <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>
So if you uncommented that export line you would experience the behaviour you described. 
Assuming you have not done that, could it be that Bria is simply rejecting any INVITE with SDP that does not contain an SAVP entry with a crypto attribute? If this was the case you would find all inbound call to that extension failing.

Actually I wonder if this is what happened and then caused you to uncomment the line above, which has led you to your conclusion, as this would cause only calls coming from SRTP devices to work. If so, you'd want to comment it out again and find a different way to create a group for all users with SRTP devices and use a dialplan condition to decide whether or not to export sip_secure_media=true. 

Alternatively you could try for some sort of fall-back mechanism but you'd have to think carefully about this to make it secure and/or stable.
Good luck!

Richard
PS: In your first paragraph, did you mean Bria for iPhone in both cases?


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

Join Us At ClueCon - Aug 7-9, 2012

FreeSWITCH-users mailing 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 rdmitry0911 at gmail.com  Mon Aug 27 13:48:45 2012
From: rdmitry0911 at gmail.com (Dmitry R)
Date: Mon, 27 Aug 2012 13:48:45 +0400
Subject: [Freeswitch-users] Google Voice doesn't work any more
In-Reply-To: <CAKV-jFwqp9R=j2bEv7fQnL4ticFTXBeJEvjmyCyDoL3wJXUt=w@mail.gmail.com>
Message-ID: <!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAATLu3mYWl7E69+584sdKQ4sKAAAAQAAAAa1jzGHgnDk+5iXo4jvBrggEAAAAA@gmail.com>

Does anybody know what happened to Google Voice? It doesn't work with FS
last several days. 2 weeks ago everything was OK. Now I see errors in
dingaling debug output like this:
 
2012-08-27 10:28:19.902527 [INFO] libdingaling.c:1743 SecRECV:
----------------------------------------------------------------------------
---
<iq from="+1xxxxxxxxxx at voice.google.com/srvres-MTAuMTc2LjQ1LjIyOjk4MDQ="
to="xxxxxxxx at gmail.com/talkB28037C5 <mailto:xxxxxxxx at gmail.com/talkB28037C5>
" id="jingle:10.176.45.22-25598321:1:4B61C7FB" type="set">
  <ses:session type="terminate" id="7084185241"
initiator="xxxxxxxxxx at gmail.com/talkB28037C5
<mailto:dxxxxxxxxxx at gmail.com/talkB28037C5> "
xmlns:ses="http://www.google.com/session">
    <ses:reason>
      <ses:general-error></ses:general-error>
      <ses:bad-request></ses:bad-request>
      <ses:text>Error handling stanza</ses:text>
    </ses:reason>
    <pho:internal-server-error
xmlns:pho="http://www.google.com/session/phone"
<http://www.google.com/session/phone> ></pho:internal-server-error>
  </ses:session>
</iq>
 
2012-08-27 10:28:20.002533 [INFO] libdingaling.c:1743 SecRECV:
----------------------------------------------------------------------------
---
<iq from="+1xxxxxxxxxx at voice.google.com/srvres-MTAuMTc2LjQ1LjIyOjk4MDQ="
to="xxxxxxxx at gmail.com/talkB28037C5
<mailto:dxxxxxxxx at gmail.com/talkB28037C5> " type="error" id="307">
  <ses:session type="initiate" id="7084185241"
initiator="xxxxxxxx at gmail.com/talkB28037C5
<mailto:dxxxxxxxx at gmail.com/talkB28037C5> "
xmlns:ses="http://www.google.com/session">
    <vid:description xmlns:vid="http://www.google.com/session/video">
      <vid:payload-type id="97" name="H264"></vid:payload-type>
      <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000"
xmlns:pho="http://www.google.com/session/phone"
<http://www.google.com/session/phone> ></pho:payload-type>
    </vid:description>
  </ses:session>
  <error code="400" type="modify">
    <bad-request xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></bad-request>
    <text xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">Missing phone
description</text>
  </error>
</iq>

 
2012-08-27 10:28:20.222524 [INFO] libdingaling.c:1743 SecRECV:
----------------------------------------------------------------------------
---
<iq from="+1xxxxxxxxxx at voice.google.com/srvres-MTAuMTc2LjQ1LjIyOjk4MDQ="
to="xxxxxxxx at gmail.com/talkB28037C5
<mailto:dxxxxxxxx at gmail.com/talkB28037C5> " type="error" id="308">
  <ses:session type="terminate" id="7084185241"
initiator="xxxxxxxx at gmail.com/talkB28037C5
<mailto:dxxxxxxxx at gmail.com/talkB28037C5> "
xmlns:ses="http://www.google.com/session" <http://www.google.com/session>
></ses:session>
  <error code="404" type="cancel">
    <item-not-found
xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></item-not-found>
    <text xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">No such session</text>
  </error>
</iq>
 
Google Voice client profile is exactly like described in wiki
<http://wiki.freeswitch.org/wiki/Google_Voice>
http://wiki.freeswitch.org/wiki/Google_Voice 
 
At the same time iPhone application Talkatone which is makes calls via
Google Voice works great with the same creedentials.
Any idea of what is wrong would be very much appreciated
 
Thank you, Dmitry
 
 

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From govoiper at gmail.com  Mon Aug 27 14:54:34 2012
From: govoiper at gmail.com (SamyGo)
Date: Mon, 27 Aug 2012 15:54:34 +0500
Subject: [Freeswitch-users] How to know inbound call arrive from which
	gateway?
In-Reply-To: <BLU0-SMTP4659A4A0496E38F9F7E779D8CA20@phx.gbl>
References: <BLU0-SMTP4659A4A0496E38F9F7E779D8CA20@phx.gbl>
Message-ID: <CAJUJwtjg5kV2La9hRKQiNQhH-kmbXvVKhP8MGhmaOpQNSZwrUQ@mail.gmail.com>

Hi,
What you can do easily is just change the context of each gateway and then
in that dialplan code append a prefix to the destination_number received.
i.e save destination_number in another variable and append the desired
prefix to that variable. Use the new variable anywhere in your dialplan as
you want !

BR
Sammy


On Mon, Aug 27, 2012 at 9:47 AM, Pip Live <itispip-qq at hotmail.com> wrote:

> Sorry this maybe a very simple question but I checked Wiki/Dialplan
> without finding an answer.
>
> How can I know inbound call arrive at which sofia gateway in dialplan? My
> purpose is to add a prefix to all inbound call from a certain gateway. I
> cannot use the "destination_number" built_in variable for this purpose, as
> in my case DID bind to a specific gateway hardware is often changing. Is
> that another varialbe to be used, eg. a gateway name etc?
>
> /Alex
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-list at puzzled.xs4all.nl  Mon Aug 27 15:03:39 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Mon, 27 Aug 2012 13:03:39 +0200
Subject: [Freeswitch-users] How to know inbound call arrive from which
 gateway?
In-Reply-To: <BLU0-SMTP4659A4A0496E38F9F7E779D8CA20@phx.gbl>
References: <BLU0-SMTP4659A4A0496E38F9F7E779D8CA20@phx.gbl>
Message-ID: <503B540B.1060909@puzzled.xs4all.nl>

On 27-08-12 06:47, Pip Live wrote:
> Sorry this maybe a very simple question but I checked Wiki/Dialplan
> without finding an answer.
>
> How can I know inbound call arrive at which sofia gateway in dialplan?
> My purpose is to add a prefix to all inbound call from a certain
> gateway. I cannot use the "destination_number" built_in variable for
> this purpose, as in my case DID bind to a specific gateway hardware is
> often changing. Is that another varialbe to be used, eg. a gateway name etc?

Until someone comes up with a better answer, have you tried running the 
info app on the incoming call? It gives a ton of info about the channel.

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

Regards,
Patrick



From avi at avimarcus.net  Mon Aug 27 15:21:17 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 27 Aug 2012 14:21:17 +0300
Subject: [Freeswitch-users] How to know inbound call arrive from which
	gateway?
In-Reply-To: <503B540B.1060909@puzzled.xs4all.nl>
References: <BLU0-SMTP4659A4A0496E38F9F7E779D8CA20@phx.gbl>
	<503B540B.1060909@puzzled.xs4all.nl>
Message-ID: <CALNZuwmUjm8S76sV1qg31=g3y8SN7KVgzoSqRYwePjx5rA94Yw@mail.gmail.com>

A separate context for each sounds messy.
Indeed, try INFO, but I don't think we actually have a variable for the
gateway.
However, you can *set* a channel variable on inbound calls on the gateway:
http://wiki.freeswitch.org/wiki/Clarification:gateways#Variables

e.g.<variables> <variable name="gateway" data="flowroute"
direction="inbound"/></variables>

-Avi



On Mon, Aug 27, 2012 at 2:03 PM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 27-08-12 06:47, Pip Live wrote:
> > Sorry this maybe a very simple question but I checked Wiki/Dialplan
> > without finding an answer.
> >
> > How can I know inbound call arrive at which sofia gateway in dialplan?
> > My purpose is to add a prefix to all inbound call from a certain
> > gateway. I cannot use the "destination_number" built_in variable for
> > this purpose, as in my case DID bind to a specific gateway hardware is
> > often changing. Is that another varialbe to be used, eg. a gateway name
> etc?
>
> Until someone comes up with a better answer, have you tried running the
> info app on the incoming call? It gives a ton of info about the channel.
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_info
>
> Regards,
> Patrick
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 w8hdkim at gmail.com  Mon Aug 27 15:45:19 2012
From: w8hdkim at gmail.com (Kim Culhan)
Date: Mon, 27 Aug 2012 07:45:19 -0400
Subject: [Freeswitch-users] Polycom Registration Failing
In-Reply-To: <CAKZxVQVQLTZMa39VAN3cwEBYXCHuRsmpu0Y=Edh28=cHKm134A@mail.gmail.com>
References: <CAKZxVQVQLTZMa39VAN3cwEBYXCHuRsmpu0Y=Edh28=cHKm134A@mail.gmail.com>
Message-ID: <CAKZxVQVroZPs_d6msrD0+2d4rcphf9AkGgLA+9UQSf6o9ocgLQ@mail.gmail.com>

On Sun, August 26, 2012 7:41 pm, Brian Foster wrote:
> I have a Polycom IP 335 and a Linksys SPA921 here at my office (on the same
> desk, connected to the same hosted PBX; don't ask). The Linksys is not
> having issues registering. The Polycom, however, gets a 401 Unauthorized
> every time it tries to register. Both phones are behind the same NAT, both
> are registering to the same PBX, and they are registering to the same
> extension.
>
> SIP messages here: http://pastebin.freeswitch.org/19765
> <http://pastebin.freeswitch.org/19765>Internal Sofia Profile here:
> http://pastebin.freeswitch.org/19766
>
> Any hints? I've reset both phones, updated to latest in v1.2.stable, etc.
> Still no joy.

For Polycom phones, in the Internal Profile:

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

-kim


From itispip-qq at hotmail.com  Mon Aug 27 15:55:25 2012
From: itispip-qq at hotmail.com (Pip Live)
Date: Mon, 27 Aug 2012 19:55:25 +0800
Subject: [Freeswitch-users] How to know inbound call arrive from which
	gateway?
In-Reply-To: <CALNZuwmUjm8S76sV1qg31=g3y8SN7KVgzoSqRYwePjx5rA94Yw@mail.gmail.com>
References: <BLU0-SMTP4659A4A0496E38F9F7E779D8CA20@phx.gbl>
	<503B540B.1060909@puzzled.xs4all.nl>
	<CALNZuwmUjm8S76sV1qg31=g3y8SN7KVgzoSqRYwePjx5rA94Yw@mail.gmail.com>
Message-ID: <BLU0-SMTP31974619ADA2D52D1BC4B748CA20@phx.gbl>

Thanks Avi. Yours idea seems very handy & lowest cost to implement. Will
try that.

/Alex
? 2012-8-27 ??7:21?"Avi Marcus" <avi at avimarcus.net>???

> A separate context for each sounds messy.
> Indeed, try INFO, but I don't think we actually have a variable for the
> gateway.
> However, you can *set* a channel variable on inbound calls on the
> gateway:
> http://wiki.freeswitch.org/wiki/Clarification:gateways#Variables
>
> e.g.<variables> <variable name="gateway" data="flowroute"
> direction="inbound"/></variables>
>
> -Avi
>
>
>
> On Mon, Aug 27, 2012 at 2:03 PM, Patrick Lists <
> freeswitch-list at puzzled.xs4all.nl> wrote:
>
>> On 27-08-12 06:47, Pip Live wrote:
>> > Sorry this maybe a very simple question but I checked Wiki/Dialplan
>> > without finding an answer.
>> >
>> > How can I know inbound call arrive at which sofia gateway in dialplan?
>> > My purpose is to add a prefix to all inbound call from a certain
>> > gateway. I cannot use the "destination_number" built_in variable for
>> > this purpose, as in my case DID bind to a specific gateway hardware is
>> > often changing. Is that another varialbe to be used, eg. a gateway name
>> etc?
>>
>> Until someone comes up with a better answer, have you tried running the
>> info app on the incoming call? It gives a ton of info about the channel.
>>
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_info
>>
>> Regards,
>> Patrick
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nickolayr at gmail.com  Mon Aug 27 17:03:55 2012
From: nickolayr at gmail.com (Nikolay Rogoshchenkov)
Date: Mon, 27 Aug 2012 09:03:55 -0400
Subject: [Freeswitch-users] Google Voice doesn't work any more
In-Reply-To: <!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAATLu3mYWl7E69+584sdKQ4sKAAAAQAAAAa1jzGHgnDk+5iXo4jvBrggEAAAAA@gmail.com>
References: <CAKV-jFwqp9R=j2bEv7fQnL4ticFTXBeJEvjmyCyDoL3wJXUt=w@mail.gmail.com>
	<!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAATLu3mYWl7E69+584sdKQ4sKAAAAQAAAAa1jzGHgnDk+5iXo4jvBrggEAAAAA@gmail.com>
Message-ID: <CAAQCs30mHpYU=hg8dCTphUcRSgCtBU8F+hgfNdm6FUzfUOLruw@mail.gmail.com>

Did you try this: "*Note: You must remove h264 from
conf/autoload_configs/dingaling.conf.xml and just leave PCMU because Google
Voice doesn't support
it.<http://wiki.freeswitch.org/wiki/Google_Voice#Receiving_Calls_From_Google_Voice>"
?*


--
Rogoshchenkov Nikolay


On Mon, Aug 27, 2012 at 5:48 AM, Dmitry R <rdmitry0911 at gmail.com> wrote:

> **
> *Does anybody know what happened to Google Voice? It doesn't work with FS
> last several days. 2 weeks ago everything was OK. Now I see errors in
> dingaling debug output like this:*
> **
> 2012-08-27 10:28:19.902527 [INFO] libdingaling.c:1743 SecRECV:
>
> -------------------------------------------------------------------------------
> <iq from="+1xxxxxxxxxx at voice.google.com/srvres-MTAuMTc2LjQ1LjIyOjk4MDQ="
> to="xxxxxxxx at gmail.com/talkB28037C5 <xxxxxxxx at gmail.com/talkB28037C5>"
> id="jingle:10.176.45.22-25598321:1:4B61C7FB" type="set">
>   <ses:session type="terminate" id="7084185241" initiator="
> xxxxxxxxxx at gmail.com/talkB28037C5 <dxxxxxxxxxx at gmail.com/talkB28037C5>"
> xmlns:ses="http://www.google.com/session">
>     <ses:reason>
>       <ses:general-error></ses:general-error>
>       <ses:bad-request></ses:bad-request>
>       <ses:text>Error handling stanza</ses:text>
>     </ses:reason>
>     <pho:internal-server-error xmlns:pho="
> http://www.google.com/session/phone"></pho:internal-server-error>
>   </ses:session>
> </iq>
>
> 2012-08-27 10:28:20.002533 [INFO] libdingaling.c:1743 SecRECV:
>
> -------------------------------------------------------------------------------
> <iq from="+1xxxxxxxxxx at voice.google.com/srvres-MTAuMTc2LjQ1LjIyOjk4MDQ="
> to="xxxxxxxx at gmail.com/talkB28037C5 <dxxxxxxxx at gmail.com/talkB28037C5>"
> type="error" id="307">
>   <ses:session type="initiate" id="7084185241" initiator="
> xxxxxxxx at gmail.com/talkB28037C5 <dxxxxxxxx at gmail.com/talkB28037C5>"
> xmlns:ses="http://www.google.com/session">
>     <vid:description xmlns:vid="http://www.google.com/session/video">
>       <vid:payload-type id="97" name="H264"></vid:payload-type>
>       <pho:payload-type id="0" name="PCMU" clockrate="8000"
> bitrate="64000" xmlns:pho="
> http://www.google.com/session/phone"></pho:payload-type>
>     </vid:description>
>   </ses:session>
>   <error code="400" type="modify">
>     <bad-request xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></bad-request>
>     <text xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">Missing phone
> description</text>
>   </error>
> </iq>
>
> 2012-08-27 10:28:20.222524 [INFO] libdingaling.c:1743 SecRECV:
>
> -------------------------------------------------------------------------------
> <iq from="+1xxxxxxxxxx at voice.google.com/srvres-MTAuMTc2LjQ1LjIyOjk4MDQ="
> to="xxxxxxxx at gmail.com/talkB28037C5 <dxxxxxxxx at gmail.com/talkB28037C5>"
> type="error" id="308">
>   <ses:session type="terminate" id="7084185241" initiator="
> xxxxxxxx at gmail.com/talkB28037C5 <dxxxxxxxx at gmail.com/talkB28037C5>"
> xmlns:ses="http://www.google.com/session"></ses:session>
>   <error code="404" type="cancel">
>     <item-not-found
> xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></item-not-found>
>     <text xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">No such
> session</text>
>   </error>
> </iq>
>
> *Google Voice client profile is exactly like described in wiki **
> http://wiki.freeswitch.org/wiki/Google_Voice*<http://wiki.freeswitch.org/wiki/Google_Voice>
> * *
> **
> *At the same time iPhone application Talkatone which is makes calls via
> Google Voice works great with the same creedentials.*
> *Any idea of what is wrong would be very much appreciated*
> **
> *Thank you, Dmitry*
> **
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ntomer at newgen.co.in  Mon Aug 27 17:15:40 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Mon, 27 Aug 2012 18:45:40 +0530
Subject: [Freeswitch-users] mod_CallCenter: Agents not being logged out
Message-ID: <02ba01cd8456$110470d0$330d5270$@co.in>

Hi,

 

I've configured FreeSWITCH with 4 agents, in the following manner:

 

<agents>

    <agent name="1000 at default" type="callback"
contact="[call_timeout=10]user/1000" status="Available" max-no-answer="3"
wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />

    <agent name="1001 at default" type="callback"
contact="[origination_caller_id_name='Queue
Caller',call_timeout=10]user/1001" status="Available" max-no-answer="30"
wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />

     <agent name="1002 at default" type="callback"
contact="[call_timeout=10]user/1002" status="Available" max-no-answer="3"
wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />

     <agent name="1003 at default" type="callback"
contact="[call_timeout=10]user/1003" status="Available" max-no-answer="3"
wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />

  </agents>

  <tiers>

    <tier agent="1000 at default" queue="sales at default" level="1"
position="1"/>

    <tier agent="1001 at default" queue="support at default" level="1"
position="1"/>

    <tier agent="1003 at default" queue="sales at default" level="1"
position="1"/>

  </tiers>

 

Now I am trying to set an agent's status with this command at the console:
callcenter_config [agent set status 'Logged out'] [1000 at default]

 

This command is executed alright, but when I execute "callcenter_config
agent list", the status of 1000 at default is still shown as "available"; why
so?

 

Also please tell me what should be the syntax if I want to set status when
agent dials a number. And is there a way to set this status from outside,
like from a web page click?

 

Regards

 

Nitin


Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From eranga.erl at gmail.com  Mon Aug 27 17:12:13 2012
From: eranga.erl at gmail.com (Eranga Udesh)
Date: Mon, 27 Aug 2012 18:42:13 +0530
Subject: [Freeswitch-users] Multiple Say in Playback
Message-ID: <CALg+e6w_F7y=m=Uv+7Oz=6yxBMG0iGAMfooEW-da_kSvzwZTqg@mail.gmail.com>

Hi,

Is it possible for me to have multiple say commands in a playback like
below (which is not working btw)?

playback(say:'First part'!say:'Second Part'!say:'Third Part')

What is the correct way to do that?

Btw, I am looking to break the Text to multi-parts instead of giving it in
a single say, since I want to say one part as String, another part as
Number Counted, etc.

Thanks in advance.
- Eranga
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From vbvbrj at gmail.com  Mon Aug 27 17:17:17 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Mon, 27 Aug 2012 16:17:17 +0300
Subject: [Freeswitch-users] mod_CallCenter: Agents not being logged out
In-Reply-To: <02ba01cd8456$110470d0$330d5270$@co.in>
References: <02ba01cd8456$110470d0$330d5270$@co.in>
Message-ID: <503B735D.4070409@gmail.com>

> Now I am trying to set an agent?s status with this command at the
> console: callcenter_config [agent set status 'Logged out'] [1000 at default]
>
> This command is executed alright, but when I execute ?callcenter_config
> agent list?, the status of 1000 at default is still shown as ?available?;
> why so?

Please see carefully how the command is written. It must be:
callcenter_config agent set status [agent name] 'Logged Out'

> Also please tell me what should be the syntax if I want to set status
> when agent dials a number. And is there a way to set this status from
> outside, like from a web page click?

On wiki 
http://wiki.freeswitch.org/wiki/Mod_callcenter#Script_to_announce_members_position 
is an example for agent-login and agent-logout dialplan extension.


From vbvbrj at gmail.com  Mon Aug 27 17:19:10 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Mon, 27 Aug 2012 16:19:10 +0300
Subject: [Freeswitch-users] Multiple Say in Playback
In-Reply-To: <CALg+e6w_F7y=m=Uv+7Oz=6yxBMG0iGAMfooEW-da_kSvzwZTqg@mail.gmail.com>
References: <CALg+e6w_F7y=m=Uv+7Oz=6yxBMG0iGAMfooEW-da_kSvzwZTqg@mail.gmail.com>
Message-ID: <503B73CE.9010204@gmail.com>

On 27.08.2012 16:12, Eranga Udesh wrote:
> Hi,
>
> Is it possible for me to have multiple say commands in a playback like
> below (which is not working btw)?
>
> playback(say:'First part'!say:'Second Part'!say:'Third Part')
>
> What is the correct way to do that?
>
> Btw, I am looking to break the Text to multi-parts instead of giving it
> in a single say, since I want to say one part as String, another part as
> Number Counted, etc.

Use
playback('phrase:some-phrase')
and create the phrase macro with everything you want.



From eranga.erl at gmail.com  Mon Aug 27 18:49:29 2012
From: eranga.erl at gmail.com (Eranga Udesh)
Date: Mon, 27 Aug 2012 20:19:29 +0530
Subject: [Freeswitch-users] Multiple Say in Playback
In-Reply-To: <503B73CE.9010204@gmail.com>
References: <CALg+e6w_F7y=m=Uv+7Oz=6yxBMG0iGAMfooEW-da_kSvzwZTqg@mail.gmail.com>
	<503B73CE.9010204@gmail.com>
Message-ID: <CALg+e6yHjWtCi9Ze52DycXvUFq25iy9cMv9fZLr8cNhcRMpOBg@mail.gmail.com>

Thanks for the feedback. I'll try this.
- Eranga


On Mon, Aug 27, 2012 at 6:49 PM, Vbvbrj <vbvbrj at gmail.com> wrote:

> On 27.08.2012 16:12, Eranga Udesh wrote:
> > Hi,
> >
> > Is it possible for me to have multiple say commands in a playback like
> > below (which is not working btw)?
> >
> > playback(say:'First part'!say:'Second Part'!say:'Third Part')
> >
> > What is the correct way to do that?
> >
> > Btw, I am looking to break the Text to multi-parts instead of giving it
> > in a single say, since I want to say one part as String, another part as
> > Number Counted, etc.
>
> Use
> playback('phrase:some-phrase')
> and create the phrase macro with everything you want.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 abaci64 at gmail.com  Mon Aug 27 19:16:23 2012
From: abaci64 at gmail.com (Abaci)
Date: Mon, 27 Aug 2012 11:16:23 -0400
Subject: [Freeswitch-users] need help on gateway
In-Reply-To: <CAMBhhBLG82V1AngMvgu1fckk8tqLTqO_5yhqopiw5=E-bU7FbA@mail.gmail.com>
References: <CAMBhhBK6sgPckoxa8JZ1Tpe-60+K-imaWXfc+7i0jHWo=RfteQ@mail.gmail.com>
	<CAKV-jFzcmu29-1LmvpU0MbqJER31jQzzJFy9X4AK6SDytSvLKg@mail.gmail.com>
	<CAMBhhBLdow__4Dk_ymVsxp4PoQTYhQ1WHD4c3qC58p6OHqnUhw@mail.gmail.com>
	<CAMBhhBLG82V1AngMvgu1fckk8tqLTqO_5yhqopiw5=E-bU7FbA@mail.gmail.com>
Message-ID: <503B8F47.1070604@gmail.com>

You miss some tags in your XML, since it's in the directory you need to 
have the following tags: "domain", "users", "user" & "gateways"
see http://wiki.freeswitch.org/wiki/Mod_xml_curl#Startup and 
http://wiki.freeswitch.org/wiki/Clarification:gateways#conf.2Fdirectory.2Fdefault.2Fexample.com.xml

On 8/27/2012 3:07 AM, haloha wrote:
> Hi
>
> i did cli in freeswitch mode: sofia profile external rescan
> and freeswitch sends to web server
>
> hostname=localhost.localdomain
> section=directory
> tag_name=
> key_name=
> key_value=
> Event-Name=REQUEST_PARAMS
> Core-UUID=4144f203-0c8f-4050-b58b-a39564428f3a
> FreeSWITCH-Hostname=localhost.localdomain
> FreeSWITCH-Switchname=localhost.localdomain
> FreeSWITCH-IPv4=172.30.41.148
> FreeSWITCH-IPv6=%3A%3A1
> Event-Date-Local=2012-08-27%2003%3A00%3A19
> Event-Date-GMT=Mon,%2027%20Aug%202012%2007%3A00%3A19%20GMT
> Event-Date-Timestamp=1346050819261656
> Event-Calling-File=sofia.c
> Event-Calling-Function=launch_sofia_worker_thread
> Event-Calling-Line-Number=2006
> Event-Sequence=431
> purpose=gateways
> profile=external
>
>
>
> i try to send back
>
> <?xml version="1.0" encoding="utf-8"?>
> <document type="freeswitch/xml">
>    <section name="directory">
>     <include>
>      <gateway name="example">
>       <param name="username" value="cluecon"/>
>       <param name="password" value="2007"/>
>       <param name="proxy" value="sip.abc.net"/>
>       <param name="expire-seconds" value="3600"/>
>       <param name="register" value="false"/>
>       <param name="retry-seconds" value="30"/>
>       <param name="ping" value="25"/>
>      </gateway>
>     </include>
>    </section>
> </document>
>
> but it does not work
>
> what is the correct format form
>
> Thank you
> Ha`
>
>
>
>
>
> On 8/27/12, haloha <haloha201 at gmail.com> wrote:
>> hi Brian
>>
>> could you please give me the example of the respond for gateway's info
>> from web server
>> there is no example for gateway's info respond,
>> i only found the respond for user's info:
>> <document type="freeswitch/xml">
>>    <section name="directory">
>>      <domain name="domain1.awesomevoipdomain.faketld">
>>        <params>
>>          <param name="dial-string"
>> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>        </params>
>>        <groups>
>>          <group name="default">
>>           <users>
>>            <user id="1004">
>>              <params>
>>                <param name="password" value="some_password"/>
>>              </params>
>>            </user>
>>           </users>
>>          </group>
>>        </groups>
>>      </domain>
>>    </section>
>> </document>
>>
>> Thank you
>> Ha`
>>
>> On 8/27/12, Brian Foster <bdfoster at endigotech.com> wrote:
>>> Easiest way is to use xml curl, checkout mod_xml_curl. You can pull
>>> things
>>> from mysql via your scripts on the web server.
>>>
>>> -BDF
>>>
>>> On Sun, Aug 26, 2012 at 11:20 PM, haloha <haloha201 at gmail.com> wrote:
>>>
>>>> Hi all
>>>>
>>>> for now, all the gateway's information is stored in xml file and its
>>>> path(conf/sip_profiles/external/example.xml):
>>>> <include>
>>>>            <gateway name="example">
>>>>                    <param name="username" value="cluecon"/>
>>>>                    <param name="password" value="2007"/>
>>>>                    <param name="proxy" value="sip.abc.net"/>
>>>>                    <param name="expire-seconds" value="3600"/>
>>>>                    <param name="register" value="true"/>
>>>>                    <!--<param name="register" value="false"/>-->
>>>>                    <param name="retry-seconds" value="30"/>
>>>>                    <param name="ping" value="25"/>
>>>>            </gateway>
>>>> </include>
>>>>
>>>> is there a way to store the gateway's information in Mysql Databases,
>>>> like sip clients information in folder conf/directory/
>>>>
>>>> Thank you
>>>> Ha`
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>
>>>
>>> --
>>> Brian D. Foster
>>> Endigo Computer LLC
>>> Email: bdfoster at endigotech.com
>>> Phone: 317-800-7876
>>> Indianapolis, Indiana, USA
>>>
>>> This message contains confidential information and is intended for those
>>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header.
>>> If
>>> you are not the intended recipient you are notified that disclosing,
>>> copying, distributing or taking any action in reliance on the contents of
>>> this information is strictly prohibited. E-mail transmission cannot be
>>> guaranteed to be secure or error-free as information could be
>>> intercepted,
>>> corrupted, lost, destroyed, arrive late or incomplete, or contain
>>> viruses.
>>> The sender therefore does not accept liability for any errors or
>>> omissions
>>> in the contents of this message, which arise as a result of e-mail
>>> transmission. If verification is required please request a hard-copy
>>> version.
>>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 gvvsubhashkumar at gmail.com  Mon Aug 27 19:19:44 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Mon, 27 Aug 2012 20:49:44 +0530
Subject: [Freeswitch-users] How to write an Application to control freeSWITCH
Message-ID: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>

Hi All,

       I am new to freeSWITCH platform.

      I want to write an application using c++ to control freeSWITCH so
please guide me.

      Thanks in advance.

Thanks,
Subhash.
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From mitch.capper at gmail.com  Mon Aug 27 19:53:41 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 27 Aug 2012 08:53:41 -0700
Subject: [Freeswitch-users] How to write an Application to control
	freeSWITCH
In-Reply-To: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
Message-ID: <CALn233py3x9eJgXAvsE6q4dL85i4ho7cJU9aF9y5Z2sit1Q_TQ@mail.gmail.com>

look at the wiki for esl probably the easiest way if you need to you
can also write fs modules in c++.
~mitch

On Mon, Aug 27, 2012 at 8:19 AM, Subhash <gvvsubhashkumar at gmail.com> wrote:
>
> Hi All,
>
>        I am new to freeSWITCH platform.
>
>       I want to write an application using c++ to control freeSWITCH so
> please guide me.
>
>       Thanks in advance.
>
> Thanks,
> Subhash.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mike at jerris.com  Mon Aug 27 20:27:21 2012
From: mike at jerris.com (Michael Jerris)
Date: Mon, 27 Aug 2012 12:27:21 -0400
Subject: [Freeswitch-users] Polycom Registration Failing
In-Reply-To: <CAKZxVQVroZPs_d6msrD0+2d4rcphf9AkGgLA+9UQSf6o9ocgLQ@mail.gmail.com>
References: <CAKZxVQVQLTZMa39VAN3cwEBYXCHuRsmpu0Y=Edh28=cHKm134A@mail.gmail.com>
	<CAKZxVQVroZPs_d6msrD0+2d4rcphf9AkGgLA+9UQSf6o9ocgLQ@mail.gmail.com>
Message-ID: <464783D1-2A12-4699-9CA8-0E651BAFB0A9@jerris.com>

This setting can cause some problems.

try setting to "safe" instead of "true"


On Aug 27, 2012, at 7:45 AM, Kim Culhan <w8hdkim at gmail.com> wrote:

> On Sun, August 26, 2012 7:41 pm, Brian Foster wrote:
>> I have a Polycom IP 335 and a Linksys SPA921 here at my office (on the same
>> desk, connected to the same hosted PBX; don't ask). The Linksys is not
>> having issues registering. The Polycom, however, gets a 401 Unauthorized
>> every time it tries to register. Both phones are behind the same NAT, both
>> are registering to the same PBX, and they are registering to the same
>> extension.
>> 
>> SIP messages here: http://pastebin.freeswitch.org/19765
>> <http://pastebin.freeswitch.org/19765>Internal Sofia Profile here:
>> http://pastebin.freeswitch.org/19766
>> 
>> Any hints? I've reset both phones, updated to latest in v1.2.stable, etc.
>> Still no joy.
> 
> For Polycom phones, in the Internal Profile:
> 
> <param name="NDLB-force-rport" value="true"/>
> 
> -kim



From ben at langfeld.co.uk  Mon Aug 27 20:41:47 2012
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Mon, 27 Aug 2012 17:41:47 +0100
Subject: [Freeswitch-users] Multiple Say in Playback
In-Reply-To: <CALg+e6yHjWtCi9Ze52DycXvUFq25iy9cMv9fZLr8cNhcRMpOBg@mail.gmail.com>
References: <CALg+e6w_F7y=m=Uv+7Oz=6yxBMG0iGAMfooEW-da_kSvzwZTqg@mail.gmail.com>
	<503B73CE.9010204@gmail.com>
	<CALg+e6yHjWtCi9Ze52DycXvUFq25iy9cMv9fZLr8cNhcRMpOBg@mail.gmail.com>
Message-ID: <CAAyX+KHi+9d3DrQ-tzDOdvx==D967oWYe5qEZQYQX5vMxOiFsg@mail.gmail.com>

Or if you're using a real TTS engine, prepare an SSML document for it to
render.

Regards,
Ben Langfeld


On 27 August 2012 15:49, Eranga Udesh <eranga.erl at gmail.com> wrote:

> Thanks for the feedback. I'll try this.
> - Eranga
>
>
> On Mon, Aug 27, 2012 at 6:49 PM, Vbvbrj <vbvbrj at gmail.com> wrote:
>
>> On 27.08.2012 16:12, Eranga Udesh wrote:
>> > Hi,
>> >
>> > Is it possible for me to have multiple say commands in a playback like
>> > below (which is not working btw)?
>> >
>> > playback(say:'First part'!say:'Second Part'!say:'Third Part')
>> >
>> > What is the correct way to do that?
>> >
>> > Btw, I am looking to break the Text to multi-parts instead of giving it
>> > in a single say, since I want to say one part as String, another part as
>> > Number Counted, etc.
>>
>> Use
>> playback('phrase:some-phrase')
>> and create the phrase macro with everything you want.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 gvvsubhashkumar at gmail.com  Mon Aug 27 20:42:19 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Mon, 27 Aug 2012 22:12:19 +0530
Subject: [Freeswitch-users] How to write an Application to control
	freeSWITCH
In-Reply-To: <CALn233py3x9eJgXAvsE6q4dL85i4ho7cJU9aF9y5Z2sit1Q_TQ@mail.gmail.com>
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
	<CALn233py3x9eJgXAvsE6q4dL85i4ho7cJU9aF9y5Z2sit1Q_TQ@mail.gmail.com>
Message-ID: <CAFvodxWB5iBu0_SGbn4tyRn7h1Jyofcc5kA5Dk0mAimjkNKAnw@mail.gmail.com>

Hi,

Thanks for the reply .if you dont mind can you point out the url to get
there because I tried but did not get needed information.

Thanjs,
Subhash.
On Aug 27, 2012 9:27 PM, "Mitch Capper" <mitch.capper at gmail.com> wrote:

> look at the wiki for esl probably the easiest way if you need to you
> can also write fs modules in c++.
> ~mitch
>
> On Mon, Aug 27, 2012 at 8:19 AM, Subhash <gvvsubhashkumar at gmail.com>
> wrote:
> >
> > Hi All,
> >
> >        I am new to freeSWITCH platform.
> >
> >       I want to write an application using c++ to control freeSWITCH so
> > please guide me.
> >
> >       Thanks in advance.
> >
> > Thanks,
> > Subhash.
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 rdmitry0911 at gmail.com  Mon Aug 27 21:17:39 2012
From: rdmitry0911 at gmail.com (Dmitry R)
Date: Mon, 27 Aug 2012 21:17:39 +0400
Subject: [Freeswitch-users] Google Voice doesn't work any more
In-Reply-To: <CAAQCs30mHpYU=hg8dCTphUcRSgCtBU8F+hgfNdm6FUzfUOLruw@mail.gmail.com>
Message-ID: <!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAATLu3mYWl7E69+584sdKQ4sKAAAAQAAAAc844VFyRcU654i3QD0GTugEAAAAA@gmail.com>

Thanks for the tip. It solves the problem.. 
 
Dmitry

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nikolay
Rogoshchenkov
Sent: Monday, August 27, 2012 5:04 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Google Voice doesn't work any more


Did you try this: "Note:
<http://wiki.freeswitch.org/wiki/Google_Voice#Receiving_Calls_From_Google_Vo
ice> You must remove h264 from conf/autoload_configs/dingaling.conf.xml and
just leave PCMU because Google Voice doesn't support it." ?


--
Rogoshchenkov Nikolay



On Mon, Aug 27, 2012 at 5:48 AM, Dmitry R <rdmitry0911 at gmail.com> wrote:



Does anybody know what happened to Google Voice? It doesn't work with FS
last several days. 2 weeks ago everything was OK. Now I see errors in
dingaling debug output like this:
 
2012-08-27 10:28:19.902527 [INFO] libdingaling.c:1743 SecRECV:
----------------------------------------------------------------------------
---
<iq from="+1xxxxxxxxxx at voice.google.com/srvres-MTAuMTc2LjQ1LjIyOjk4MDQ="
to="xxxxxxxx at gmail.com/talkB28037C5 <mailto:xxxxxxxx at gmail.com/talkB28037C5>
" id="jingle:10.176.45.22-25598321:1:4B61C7FB" type="set">
  <ses:session type="terminate" id="7084185241"
initiator="xxxxxxxxxx at gmail.com/talkB28037C5
<mailto:dxxxxxxxxxx at gmail.com/talkB28037C5> "
xmlns:ses="http://www.google.com/session">
    <ses:reason>
      <ses:general-error></ses:general-error>
      <ses:bad-request></ses:bad-request>
      <ses:text>Error handling stanza</ses:text>
    </ses:reason>
    <pho:internal-server-error
xmlns:pho="http://www.google.com/session/phone"
<http://www.google.com/session/phone%22%3E%3C/pho:internal-server-error>
></pho:internal-server-error>
  </ses:session>
</iq>
 
2012-08-27 10:28:20.002533 [INFO] libdingaling.c:1743 SecRECV:
----------------------------------------------------------------------------
---
<iq from="+1xxxxxxxxxx at voice.google.com/srvres-MTAuMTc2LjQ1LjIyOjk4MDQ="
to="xxxxxxxx at gmail.com/talkB28037C5
<mailto:dxxxxxxxx at gmail.com/talkB28037C5> " type="error" id="307">
  <ses:session type="initiate" id="7084185241"
initiator="xxxxxxxx at gmail.com/talkB28037C5
<mailto:dxxxxxxxx at gmail.com/talkB28037C5> "
xmlns:ses="http://www.google.com/session">
    <vid:description xmlns:vid="http://www.google.com/session/video">
      <vid:payload-type id="97" name="H264"></vid:payload-type>
      <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000"
xmlns:pho="http://www.google.com/session/phone"
<http://www.google.com/session/phone%22%3E%3C/pho:payload-type>
></pho:payload-type>
    </vid:description>
  </ses:session>
  <error code="400" type="modify">
    <bad-request xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></bad-request>
    <text xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">Missing phone
description</text>
  </error>
</iq>

 
2012-08-27 10:28:20.222524 [INFO] libdingaling.c:1743 SecRECV:
----------------------------------------------------------------------------
---
<iq from="+1xxxxxxxxxx at voice.google.com/srvres-MTAuMTc2LjQ1LjIyOjk4MDQ="
to="xxxxxxxx at gmail.com/talkB28037C5
<mailto:dxxxxxxxx at gmail.com/talkB28037C5> " type="error" id="308">
  <ses:session type="terminate" id="7084185241"
initiator="xxxxxxxx at gmail.com/talkB28037C5
<mailto:dxxxxxxxx at gmail.com/talkB28037C5> "
xmlns:ses="http://www.google.com/session"
<http://www.google.com/session%22%3E%3C/ses:session> ></ses:session>
  <error code="404" type="cancel">
    <item-not-found
xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></item-not-found>
    <text xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">No such session</text>
  </error>
</iq>
 
Google Voice client profile is exactly like described in wiki
<http://wiki.freeswitch.org/wiki/Google_Voice>
http://wiki.freeswitch.org/wiki/Google_Voice 
 
At the same time iPhone application Talkatone which is makes calls via
Google Voice works great with the same creedentials.
Any idea of what is wrong would be very much appreciated
 
Thank you, Dmitry
 
 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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
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From trever.adams at gmail.com  Mon Aug 27 21:49:17 2012
From: trever.adams at gmail.com (Trever L. Adams)
Date: Mon, 27 Aug 2012 11:49:17 -0600
Subject: [Freeswitch-users] Digit Echo with POTS internal and external
Message-ID: <503BB31D.10703@gmail.com>

Hello Everyone,

I am sure I am just failing to search for the right strings as this has
to be seen by others. I have FXS/FXO with no sip. Everything is POTS.

I have echo cancellation working, but I get an "echo" of DTMF digits
pressed when going from internal to external or vice versa (I am not
seeing it on internal only).

My guess is FreeSWITCH is detecting the DTMF, not blocking the already
created fast enough, and then regenerating the DTMF, so the outside IVR
or what not sees the digits twice or more.

Is there some option I am missing that will fix this?

Thank you,
Trever

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From msc at freeswitch.org  Mon Aug 27 22:16:58 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 27 Aug 2012 11:16:58 -0700
Subject: [Freeswitch-users] How to write an Application to control
	freeSWITCH
In-Reply-To: <CAFvodxWB5iBu0_SGbn4tyRn7h1Jyofcc5kA5Dk0mAimjkNKAnw@mail.gmail.com>
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
	<CALn233py3x9eJgXAvsE6q4dL85i4ho7cJU9aF9y5Z2sit1Q_TQ@mail.gmail.com>
	<CAFvodxWB5iBu0_SGbn4tyRn7h1Jyofcc5kA5Dk0mAimjkNKAnw@mail.gmail.com>
Message-ID: <CAKzWOxVhM7Qv4LqOsTRWTUugjpAxeJoB6ep9SonfTHm-nNiPMA@mail.gmail.com>

On Mon, Aug 27, 2012 at 9:42 AM, Subhash <gvvsubhashkumar at gmail.com> wrote:

> Hi,
>
> Thanks for the reply .if you dont mind can you point out the url to get
> there because I tried but did not get needed information.
>
> Thanjs,
> Subhash.
>

Subhash,

I hope you're ready to do some homework! :) When you say "control
FreeSWITCH" that can mean different things:
* Configuring FreeSWITCH
* Watching FreeSWITCH to see what calls/activity are currently taking place
* Generating new calls (outbound IVR, e.g.)
* Handling inbound calls (e.g. IVR/Auto Attendant)

Could you offer some more details about what you are trying to do with
FreeSWITCH? That will help us know whether or not ESL (event socket
library) is the right thing to recommend.

Thanks,
Michael
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From w8hdkim at gmail.com  Mon Aug 27 21:41:18 2012
From: w8hdkim at gmail.com (Kim Culhan)
Date: Mon, 27 Aug 2012 13:41:18 -0400
Subject: [Freeswitch-users] Polycom Registration Failing
Message-ID: <CAKZxVQUti88+Ez0KkXTWpzNkjNJsE7kFw9=E5hCGgSTZhNtGBA@mail.gmail.com>

On Mon, August 27, 2012 12:27 pm, Michael Jerris wrote:
> On Aug 27, 2012, at 7:45 AM, Kim Culhan <w8hdkim at gmail.com> wrote:
>
>> On Sun, August 26, 2012 7:41 pm, Brian Foster wrote:
>>> I have a Polycom IP 335 and a Linksys SPA921 here at my office (on the same
>>> desk, connected to the same hosted PBX; don't ask). The Linksys is not
>>> having issues registering. The Polycom, however, gets a 401 Unauthorized
>>> every time it tries to register. Both phones are behind the same NAT, both
>>> are registering to the same PBX, and they are registering to the same
>>> extension.
>>>
>>> SIP messages here: http://pastebin.freeswitch.org/19765
>>> <http://pastebin.freeswitch.org/19765>Internal Sofia Profile here:
>>> http://pastebin.freeswitch.org/19766
>>>
>>> Any hints? I've reset both phones, updated to latest in v1.2.stable, etc.
>>> Still no joy.
>>
>> For Polycom phones, in the Internal Profile:
>>
>> <param name="NDLB-force-rport" value="true"/>

On Mon, August 27, 2012 12:27 pm, Michael Jerris wrote:
>
> This setting can cause some problems.
>
> try setting to "safe" instead of "true"

Works for me:

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

thanks Michael

-kim


From tnsampaio at bsd.com.br  Mon Aug 27 21:55:23 2012
From: tnsampaio at bsd.com.br (Tiago N. Sampaio)
Date: Mon, 27 Aug 2012 14:55:23 -0300
Subject: [Freeswitch-users] Ringing after queue exit
Message-ID: <503BB48B.1020404@bsd.com.br>

Hi all,

     Im very new to FS, im comming from asterisk world hehe! Today i put 
my 1st FS in operation!!!

     Lets go, i have some queues in my company (queue1 queue2, etc..)
     I have max-wait on all queues to 60 seconds, and after that time i 
wanna bridge caller with some user, ex:

      <action application="callcenter" data="queue1 at default" />
      <action application="bridge" data="[leg_timeout=15]user/1002" />
      <action application="voicemail" data="default $${domain} 1002"/>

     But when "max-wait-time" arrive, and fs try bridge to user/1002, no 
ringing or moh playing, and callers think that call was droped..
     Only when voicemail is launched caller hear the sound of voicemail 
announce...

     Simplifying my question, how to put ringing on line when calling 
user/1002 after unsucessfull join on queue1?

Hugs
Tiago N. Sampaio







From abaci64 at gmail.com  Mon Aug 27 22:40:06 2012
From: abaci64 at gmail.com (Abaci)
Date: Mon, 27 Aug 2012 14:40:06 -0400
Subject: [Freeswitch-users] Ringing after queue exit
In-Reply-To: <503BB48B.1020404@bsd.com.br>
References: <503BB48B.1020404@bsd.com.br>
Message-ID: <503BBF06.8000306@gmail.com>

did you try setting transfer_rigback? 
http://wiki.freeswitch.org/wiki/Variable_transfer_ringback
On 8/27/2012 1:55 PM, Tiago N. Sampaio wrote:
> Hi all,
>
>       Im very new to FS, im comming from asterisk world hehe! Today i put
> my 1st FS in operation!!!
>
>       Lets go, i have some queues in my company (queue1 queue2, etc..)
>       I have max-wait on all queues to 60 seconds, and after that time i
> wanna bridge caller with some user, ex:
>
>        <action application="callcenter" data="queue1 at default" />
>        <action application="bridge" data="[leg_timeout=15]user/1002" />
>        <action application="voicemail" data="default $${domain} 1002"/>
>
>       But when "max-wait-time" arrive, and fs try bridge to user/1002, no
> ringing or moh playing, and callers think that call was droped..
>       Only when voicemail is launched caller hear the sound of voicemail
> announce...
>
>       Simplifying my question, how to put ringing on line when calling
> user/1002 after unsucessfull join on queue1?
>
> Hugs
> Tiago N. Sampaio
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 tahir at ictinnovations.com  Mon Aug 27 23:43:58 2012
From: tahir at ictinnovations.com (tahir almas)
Date: Tue, 28 Aug 2012 00:43:58 +0500
Subject: [Freeswitch-users] Open Source unified autodialer software
	released
In-Reply-To: <CAOL0t+JDV4cL4hZCPVLf49btScfyuVFVVyPWr8NBbnE8OpFU-Q@mail.gmail.com>
References: <CAOL0t++tKJ8_eiFRq2B7DuQ4+=xgzKRuGY+nP0a+6Paqnw+xxw@mail.gmail.com>
	<CAKbxfG82VKML36G8ovhykBAJppHO0DajXS8s5uOjQ41SOtCb0g@mail.gmail.com>
	<CAOL0t++VJoCMXTBqSYDwjNbjg1G3DV+K=j8ddpOyAZoj07HU=w@mail.gmail.com>
	<CAKbxfG81ErE-mSa6GD06A2Mv-i=qiuxSd0==62+ALV2cNHs9Zg@mail.gmail.com>
	<CAOL0t+JDV4cL4hZCPVLf49btScfyuVFVVyPWr8NBbnE8OpFU-Q@mail.gmail.com>
Message-ID: <CAOL0t+KN+6SOgz3H_XBemB_vsun9QxCzD2pJH03bDSMo5gXA0A@mail.gmail.com>

Pleased to update that both  ICTDialer http://www.ictdialer.org and ICTFAX
http://www.ictfax.org are being offered as GPL v 3.0  software

Regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT

****************************************************************************************************************
NOTICE OF CONFIDENTIALITY
This communication including any information transmitted with it is
intended only for the use of the addressees and is confidential and may be
protected by legal privilege . If you are not an intended recipient, be
aware that any disclosure, copying, distribution or use of this e-mail or
any attachment is prohibited. If you have received this e-mail in error,
please notify us immediately by returning it to the sender and delete this
copy from your system. Thank you for your cooperation.




On Wed, Jan 18, 2012 at 10:04 PM, tahir almas <tahir at ictinnovations.com>wrote:

> Thanks for valuable suggestions , will consider these recommendations
>
> Regards
> *Tahir Almas*
>
> Managing Partner
> ICT Innovations
> http://www.ictinnovations.com
> Leveraging open source in ICT
>
>
> ****************************************************************************************************************
> NOTICE OF CONFIDENTIALITY
> This communication including any information transmitted with it is
> intended only for the use of the addressees and is confidential and may
> be protected by legal privilege . If you are not an intended recipient, be
> aware that any disclosure, copying, distribution or use of this e-mail or
> any attachment is prohibited. If you have received this e-mail in error,
> please notify us immediately by returning it to the sender and delete this
> copy from your system. Thank you for your cooperation.
>
>
>
>
> On Tue, Jan 17, 2012 at 10:22 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> LGPL or GPL are ok for your case, AGPL is a little scary.   It really
>> depends if you have intent to stop anyone from using it or not, if you
>> don't then just choose MIT or BSD which are both compat with GPL and less
>> restrictive.  Remember, GPL doesn't like BSD because it says you can do
>> anything you want (including sell it or distribute it) it's hard to
>> understand at first.  GPL wants code to be free like a birdy soaring in the
>> sky not free like it costs no money,  so they actually get mad at licenses
>> with no restrictions that are considered free as in contain no restrictions
>> so you have a large political battle over weather you want it to run free
>> in a field be free of rules or be free of charge........AGPL takes it a
>> step further and says if someone downloads your project and runs it on
>> their website as a service, that it will be violating the license unless
>> they provide the code to their entire infrastructure for everyone to look
>> at.  I don't know if that will go over too well for most business ppl.
>>
>>
>> In reality debating licenses over code written in plaintext scripting
>> languages makes me chuckle a bit, just the C snob in me I suppose.   Really
>> if you choose to write code in stuff like that you may as well expect
>> people to do whatever they want with it.
>>
>> I don't even really like debating licenses at all, that's why we chose
>> BSD and MPL both the (free of restrictions) variety of license.  The only
>> obligation in MPL is to show changes to the guy who wrote the code you are
>> changing so he can see if it helps his own personal cause or not.
>>
>>
>> On Tue, Jan 17, 2012 at 12:35 AM, tahir almas <tahir at ictinnovations.com>wrote:
>>
>>> Realy thankful for your suggestion, ICTDialer is developed over Drupal
>>> 7.0  Licensed as GPL version 2 or later so we have to license ICTDialer as
>>> GPL compatible license
>>>
>>> What open source License you recommend for ICTDialer ?
>>>
>>> Regards
>>>
>>> *Tahir Almas*
>>>
>>> Managing Partner
>>> ICT Innovations
>>> http://www.ictinnovations.com
>>> Leveraging open source in ICT
>>>
>>>
>>> ****************************************************************************************************************
>>> NOTICE OF CONFIDENTIALITY
>>> This communication including any information transmitted with it is
>>> intended only for the use of the addressees and is confidential and may
>>> be protected by legal privilege . If you are not an intended recipient, be
>>> aware that any disclosure, copying, distribution or use of this e-mail or
>>> any attachment is prohibited. If you have received this e-mail in error,
>>> please notify us immediately by returning it to the sender and delete this
>>> copy from your system. Thank you for your cooperation.
>>>
>>>
>>>
>>>
>>> On Tue, Jan 17, 2012 at 6:10 AM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> One piece of advice is to not release it under AGPL which is the one
>>>> that triggers the copyleft over a socket and will turn away 99% of your
>>>> perspective testers.
>>>>
>>>>
>>>> On Mon, Jan 16, 2012 at 1:46 PM, tahir almas <tahir at ictinnovations.com>wrote:
>>>>
>>>>> Pleased to announce the release of open source Fax , SMS and Voice
>>>>> broadcasting software solution ICTDialer http://www.ictdialer.orgdeveloped over reknown  Drupal Conent Mnagment System and powerfull Plivo
>>>>> Communication framework , Your contribution and suggestions are welcome
>>>>>
>>>>> Regards
>>>>> *Tahir Almas*
>>>>>
>>>>> Managing Partner
>>>>> ICT Innovations
>>>>> http://www.ictinnovations.com
>>>>> Leveraging open source in ICT
>>>>>
>>>>>
>>>>> ****************************************************************************************************************
>>>>> NOTICE OF CONFIDENTIALITY
>>>>> This communication including any information transmitted with it is
>>>>> intended only for the use of the addressees and is confidential and
>>>>> may be protected by legal privilege . If you are not an intended recipient,
>>>>> be aware that any disclosure, copying, distribution or use of this e-mail
>>>>> or any attachment is prohibited. If you have received this e-mail in error,
>>>>> please notify us immediately by returning it to the sender and delete this
>>>>> copy from your system. Thank you for your cooperation.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 asaad2 at gmail.com  Mon Aug 27 23:46:18 2012
From: asaad2 at gmail.com (BookBag)
Date: Mon, 27 Aug 2012 15:46:18 -0400
Subject: [Freeswitch-users] forgot password on fusionpbx
Message-ID: <CAAdTMa8K+HEoaLs=ycSt+AUyrAwUkTEL=uY826jye24jHLRp=Q@mail.gmail.com>

Hello all, I accidentally typed the wrong password when setting up
fusionpbx web login. Does anyone know how I can reset it from the console
or terminal. I installed it with postgresql
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From krice at freeswitch.org  Mon Aug 27 23:53:59 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 27 Aug 2012 14:53:59 -0500
Subject: [Freeswitch-users] Open Source unified autodialer software
 released
In-Reply-To: <CAOL0t+KN+6SOgz3H_XBemB_vsun9QxCzD2pJH03bDSMo5gXA0A@mail.gmail.com>
Message-ID: <CC613A87.42A38%krice@freeswitch.org>

Why GPLv3?? GPLv3 goes against the spirit of the FreeSWITCH project in
general


On 8/27/12 2:43 PM, "tahir almas" <tahir at ictinnovations.com> wrote:

> Pleased to update that both? ICTDialer http://www.ictdialer.org and ICTFAX
> http://www.ictfax.org are being offered as GPL v 3.0? software
> 
> Regards
> Tahir Almas
> 
> Managing Partner
> ICT Innovations
> http://www.ictinnovations.com
> Leveraging open source in ICT
> 
> ******************************************************************************
> **********************************
> NOTICE OF CONFIDENTIALITY
> This communication including any information transmitted with it is intended
> only for the use of the addressees and is confidential and may be protected by
> legal privilege . If you are not an intended recipient, be aware that any
> disclosure, copying, distribution or use of this e-mail or any attachment is
> prohibited. If you have received this e-mail in error, please notify us
> immediately by returning it to the sender and delete this copy from your
> system. Thank you for your cooperation.
> ? 
> 
> 
> 
> On Wed, Jan 18, 2012 at 10:04 PM, tahir almas <tahir at ictinnovations.com>
> wrote:
>> Thanks for valuable suggestions , will consider these recommendations
>> 
>> Regards
>> Tahir Almas
>> 
>> Managing Partner
>> ICT Innovations
>> http://www.ictinnovations.com
>> Leveraging open source in ICT
>> 
>> *****************************************************************************
>> ***********************************
>> NOTICE OF CONFIDENTIALITY
>> This communication including any information transmitted with it is intended
>> only for the use of the addressees and is confidential and may be protected
>> by legal privilege . If you are not an intended recipient, be aware that any
>> disclosure, copying, distribution or use of this e-mail or any attachment is
>> prohibited. If you have received this e-mail in error, please notify us
>> immediately by returning it to the sender and delete this copy from your
>> system. Thank you for your cooperation.
>> ? 
>> 
>> 
>> 
>> On Tue, Jan 17, 2012 at 10:22 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> LGPL or GPL are ok for your case, AGPL is a little scary. ? It really
>>> depends if you have intent to stop anyone from using it or not, if you don't
>>> then just choose MIT or BSD which are both compat with GPL and less
>>> restrictive. ?Remember, GPL doesn't like BSD because it says you can do
>>> anything you want (including sell it or distribute it) it's hard to
>>> understand at first. ?GPL wants code to be free like a birdy soaring in the
>>> sky not free like it costs no money, ?so they actually get mad at licenses
>>> with no restrictions that are considered free as in contain no restrictions
>>> so you have a large political battle over weather you want it to run free in
>>> a field be free of rules or be free of charge........AGPL takes it a step
>>> further and says if someone downloads your project and runs it on their
>>> website as a service, that it will be violating the license unless they
>>> provide the code to their entire infrastructure for everyone to look at. ?I
>>> don't know if that will go over too well for most business ppl.
>>> 
>>> 
>>> In reality debating licenses over code written in plaintext scripting
>>> languages makes me chuckle a bit, just the C snob in me I suppose. ? Really
>>> if you choose to write code in stuff like that you may as well expect people
>>> to do whatever they want with it.
>>> 
>>> I don't even really like debating licenses at all, that's why we chose BSD
>>> and MPL both the (free of restrictions) variety of license. ?The only
>>> obligation in MPL is to show changes to the guy who wrote the code you are
>>> changing so he can see if it helps his own personal cause or not.
>>> 
>>> 
>>> On Tue, Jan 17, 2012 at 12:35 AM, tahir almas <tahir at ictinnovations.com>
>>> wrote:
>>>> Realy thankful for your suggestion, ICTDialer is developed over Drupal 7.0?
>>>> Licensed as GPL version 2 or later so we have to license ICTDialer as GPL
>>>> compatible license
>>>> 
>>>> What open source License you recommend for ICTDialer ?
>>>> 
>>>> Regards
>>>> 
>>>> Tahir Almas
>>>> 
>>>> Managing Partner
>>>> ICT Innovations
>>>> http://www.ictinnovations.com
>>>> Leveraging open source in ICT
>>>> 
>>>> ***************************************************************************
>>>> *************************************
>>>> NOTICE OF CONFIDENTIALITY
>>>> This communication including any information transmitted with it is
>>>> intended only for the use of the addressees and is confidential and may be
>>>> protected by legal privilege . If you are not an intended recipient, be
>>>> aware that any disclosure, copying, distribution or use of this e-mail or
>>>> any attachment is prohibited. If you have received this e-mail in error,
>>>> please notify us immediately by returning it to the sender and delete this
>>>> copy from your system. Thank you for your cooperation.
>>>> ? 
>>>> 
>>>> 
>>>> 
>>>> On Tue, Jan 17, 2012 at 6:10 AM, Anthony Minessale
>>>> <anthony.minessale at gmail.com> wrote:
>>>>> One piece of advice is to not release it under AGPL which is the one that
>>>>> triggers the copyleft over a socket and will turn away 99% of your
>>>>> perspective testers.
>>>>> 
>>>>> 
>>>>> On Mon, Jan 16, 2012 at 1:46 PM, tahir almas <tahir at ictinnovations.com>
>>>>> wrote:
>>>>>> Pleased to announce the release of open source Fax , SMS and Voice
>>>>>> broadcasting software solution ICTDialer http://www.ictdialer.org
>>>>>> developed over reknown? Drupal Conent Mnagment System and powerfull Plivo
>>>>>> Communication framework , Your contribution and suggestions are welcome
>>>>>> 
>>>>>> Regards
>>>>>> Tahir Almas
>>>>>> 
>>>>>> Managing Partner
>>>>>> ICT Innovations
>>>>>> http://www.ictinnovations.com
>>>>>> Leveraging open source in ICT
>>>>>> 
>>>>>> *************************************************************************
>>>>>> ***************************************
>>>>>> NOTICE OF CONFIDENTIALITY
>>>>>> This communication including any information transmitted with it is
>>>>>> intended only for the use of the addressees and is confidential and may
>>>>>> be protected by legal privilege . If you are not an intended recipient,
>>>>>> be aware that any disclosure, copying, distribution or use of this e-mail
>>>>>> or any attachment is prohibited. If you have received this e-mail in
>>>>>> error, please notify us immediately by returning it to the sender and
>>>>>> delete this copy from your system. Thank you for your cooperation.
>>>>>> ? 
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>> 
>>>>> 
>>>>> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From krice at freeswitch.org  Mon Aug 27 23:54:57 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 27 Aug 2012 14:54:57 -0500
Subject: [Freeswitch-users] forgot password on fusionpbx
In-Reply-To: <CAAdTMa8K+HEoaLs=ycSt+AUyrAwUkTEL=uY826jye24jHLRp=Q@mail.gmail.com>
Message-ID: <CC613AC1.42A39%krice@freeswitch.org>

This is a question for the FusionPBX guys... You should ask on their list

K


On 8/27/12 2:46 PM, "BookBag" <asaad2 at gmail.com> wrote:

> Hello all, I accidentally typed the wrong password when setting up fusionpbx
> web login. Does anyone know how I can reset it from the console or terminal. I
> installed it with postgresql
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From anthony.minessale at gmail.com  Tue Aug 28 01:25:27 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 27 Aug 2012 16:25:27 -0500
Subject: [Freeswitch-users] Open Source unified autodialer software
	released
In-Reply-To: <CC613A87.42A38%krice@freeswitch.org>
References: <CAOL0t+KN+6SOgz3H_XBemB_vsun9QxCzD2pJH03bDSMo5gXA0A@mail.gmail.com>
	<CC613A87.42A38%krice@freeswitch.org>
Message-ID: <CAKbxfG8TLHVdAdMojh_F6wQmmoz=dBGqT_sG=yvGc9F=0jpzMA@mail.gmail.com>

yah gpl v3 means "free as long as you don't use it" don't associate FS
with that kind of license scheme please.


On Mon, Aug 27, 2012 at 2:53 PM, Ken Rice <krice at freeswitch.org> wrote:
> Why GPLv3?? GPLv3 goes against the spirit of the FreeSWITCH project in
> general
>
>
>
> On 8/27/12 2:43 PM, "tahir almas" <tahir at ictinnovations.com> wrote:
>
> Pleased to update that both  ICTDialer http://www.ictdialer.org and ICTFAX
> http://www.ictfax.org are being offered as GPL v 3.0  software
>
> Regards
> Tahir Almas
>
> Managing Partner
> ICT Innovations
> http://www.ictinnovations.com
> Leveraging open source in ICT
>
> ****************************************************************************************************************
> NOTICE OF CONFIDENTIALITY
> This communication including any information transmitted with it is intended
> only for the use of the addressees and is confidential and may be protected
> by legal privilege . If you are not an intended recipient, be aware that any
> disclosure, copying, distribution or use of this e-mail or any attachment is
> prohibited. If you have received this e-mail in error, please notify us
> immediately by returning it to the sender and delete this copy from your
> system. Thank you for your cooperation.
>
>
>
>
> On Wed, Jan 18, 2012 at 10:04 PM, tahir almas <tahir at ictinnovations.com>
> wrote:
>
> Thanks for valuable suggestions , will consider these recommendations
>
> Regards
> Tahir Almas
>
> Managing Partner
> ICT Innovations
> http://www.ictinnovations.com
> Leveraging open source in ICT
>
> ****************************************************************************************************************
> NOTICE OF CONFIDENTIALITY
> This communication including any information transmitted with it is intended
> only for the use of the addressees and is confidential and may be protected
> by legal privilege . If you are not an intended recipient, be aware that any
> disclosure, copying, distribution or use of this e-mail or any attachment is
> prohibited. If you have received this e-mail in error, please notify us
> immediately by returning it to the sender and delete this copy from your
> system. Thank you for your cooperation.
>
>
>
>
> On Tue, Jan 17, 2012 at 10:22 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>
> LGPL or GPL are ok for your case, AGPL is a little scary.   It really
> depends if you have intent to stop anyone from using it or not, if you don't
> then just choose MIT or BSD which are both compat with GPL and less
> restrictive.  Remember, GPL doesn't like BSD because it says you can do
> anything you want (including sell it or distribute it) it's hard to
> understand at first.  GPL wants code to be free like a birdy soaring in the
> sky not free like it costs no money,  so they actually get mad at licenses
> with no restrictions that are considered free as in contain no restrictions
> so you have a large political battle over weather you want it to run free in
> a field be free of rules or be free of charge........AGPL takes it a step
> further and says if someone downloads your project and runs it on their
> website as a service, that it will be violating the license unless they
> provide the code to their entire infrastructure for everyone to look at.  I
> don't know if that will go over too well for most business ppl.
>
>
> In reality debating licenses over code written in plaintext scripting
> languages makes me chuckle a bit, just the C snob in me I suppose.   Really
> if you choose to write code in stuff like that you may as well expect people
> to do whatever they want with it.
>
> I don't even really like debating licenses at all, that's why we chose BSD
> and MPL both the (free of restrictions) variety of license.  The only
> obligation in MPL is to show changes to the guy who wrote the code you are
> changing so he can see if it helps his own personal cause or not.
>
>
> On Tue, Jan 17, 2012 at 12:35 AM, tahir almas <tahir at ictinnovations.com>
> wrote:
>
> Realy thankful for your suggestion, ICTDialer is developed over Drupal 7.0
> Licensed as GPL version 2 or later so we have to license ICTDialer as GPL
> compatible license
>
> What open source License you recommend for ICTDialer ?
>
> Regards
>
> Tahir Almas
>
> Managing Partner
> ICT Innovations
> http://www.ictinnovations.com
> Leveraging open source in ICT
>
> ****************************************************************************************************************
> NOTICE OF CONFIDENTIALITY
> This communication including any information transmitted with it is intended
> only for the use of the addressees and is confidential and may be protected
> by legal privilege . If you are not an intended recipient, be aware that any
> disclosure, copying, distribution or use of this e-mail or any attachment is
> prohibited. If you have received this e-mail in error, please notify us
> immediately by returning it to the sender and delete this copy from your
> system. Thank you for your cooperation.
>
>
>
>
> On Tue, Jan 17, 2012 at 6:10 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>
> One piece of advice is to not release it under AGPL which is the one that
> triggers the copyleft over a socket and will turn away 99% of your
> perspective testers.
>
>
> On Mon, Jan 16, 2012 at 1:46 PM, tahir almas <tahir at ictinnovations.com>
> wrote:
>
> Pleased to announce the release of open source Fax , SMS and Voice
> broadcasting software solution ICTDialer http://www.ictdialer.org developed
> over reknown  Drupal Conent Mnagment System and powerfull Plivo
> Communication framework , Your contribution and suggestions are welcome
>
> Regards
> Tahir Almas
>
> Managing Partner
> ICT Innovations
> http://www.ictinnovations.com
> Leveraging open source in ICT
>
> ****************************************************************************************************************
> NOTICE OF CONFIDENTIALITY
> This communication including any information transmitted with it is intended
> only for the use of the addressees and is confidential and may be protected
> by legal privilege . If you are not an intended recipient, be aware that any
> disclosure, copying, distribution or use of this e-mail or any attachment is
> prohibited. If you have received this e-mail in error, please notify us
> immediately by returning it to the sender and delete this copy from your
> system. Thank you for your cooperation.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
>
> --
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From abaci64 at gmail.com  Tue Aug 28 02:27:27 2012
From: abaci64 at gmail.com (Abaci)
Date: Mon, 27 Aug 2012 18:27:27 -0400
Subject: [Freeswitch-users] voicemail to email not working
Message-ID: <503BF44F.6050007@gmail.com>

I'm trying to get FreeSWITCH to send my voicemail to my email and can't 
it to work. I have sSMTP installed and working (tested using 
voicemail.tpl) I see in the FreeSWITCH logs *"switch_utils.c:761 Emailed 
file [/tmp/mail.13461330724a96] to [me at mydomain.com]" *however in 
/var/log/maillog I see nothing and I don't get the email. does someone 
have a clue on what might be wrong here, or is there any debug I can 
enable for this?

_*in switch.conf.xml I have:*_
     <param name="mailer-app" value="sendmail"/>
     <param name="mailer-app-args" value="-t"/>

_*in voicemail.conf.xml I have:*_
     <email>
         <param name="template-file" value="voicemail.tpl"/>
         <param name="notify-template-file" value="notify-voicemail.tpl"/>
         <!-- this is the format voicemail_time will have -->
         <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
         <!--<param name="email-from" 
value="${voicemail_account}@${voicemail_domain}"/>-->
     </email>

_*in the directory (xml_curl) I have:*_
<document type="freeswitch/xml">
   <section name="directory">
     <domain name="mydomain.com">
       <user id="200">
         <params>
           <param name="password" value="cj987r5y95"/>
           <param name="vm-password" value="1234"/>
           <param name="vm-enabled" value="true"/>
           <param name="vm-email-all-messages" value="true"/>
           <param name="vm-mailto" value="me at mydomain.com"/>
           <param name="vm-mailfrom" value="Voicemail Server 
<voicemail at mydomain.com>"/>
           <param name="vm-attach-file" value="true" />
           <param name="dial-string" 
value="${sofia_contact(200 at mydomain.com)}"/>
         </params>
         <variables>
           <variable name="toll_allow" 
value="domestic,international,local"/>
           <variable name="user_context" value="routing"/>
           <variable name="effective_caller_id_name" value="test user"/>
           <variable name="effective_caller_id_number" value="2125551212"/>
           <variable name="outbound_caller_id_number" value="2125551212"/>
         </variables>
       </user>
     </domain>
   </section>
</document>

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From gvvsubhashkumar at gmail.com  Tue Aug 28 07:14:43 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Tue, 28 Aug 2012 08:44:43 +0530
Subject: [Freeswitch-users] How to write an Application to control
	freeSWITCH
In-Reply-To: <CAKzWOxVhM7Qv4LqOsTRWTUugjpAxeJoB6ep9SonfTHm-nNiPMA@mail.gmail.com>
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
	<CALn233py3x9eJgXAvsE6q4dL85i4ho7cJU9aF9y5Z2sit1Q_TQ@mail.gmail.com>
	<CAFvodxWB5iBu0_SGbn4tyRn7h1Jyofcc5kA5Dk0mAimjkNKAnw@mail.gmail.com>
	<CAKzWOxVhM7Qv4LqOsTRWTUugjpAxeJoB6ep9SonfTHm-nNiPMA@mail.gmail.com>
Message-ID: <CAFvodxW6435KvLQHUEkSiq+BB+BpE6fdF_mRv17eMy_AXEKzog@mail.gmail.com>

Michael,

           You are right we are more concern about the below things but i
would like to add more things in your list. Please find my comments in red.

    * Configuring FreeSWITCH(this include all conf files and dialplan
generation ??)
    * Watching FreeSWITCH to see what calls/activity are currently taking
place
    * Generating new calls (outbound IVR, e.g.)
    * Handling inbound calls (e.g. IVR/Auto Attendant).
    * How to use freeswitch for billing purpose.
    *Bridging and Transferring the inbound calls.



Thanks,
Subhash.


On Mon, Aug 27, 2012 at 11:46 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Mon, Aug 27, 2012 at 9:42 AM, Subhash <gvvsubhashkumar at gmail.com>wrote:
>
>> Hi,
>>
>> Thanks for the reply .if you dont mind can you point out the url to get
>> there because I tried but did not get needed information.
>>
>> Thanjs,
>> Subhash.
>>
>
> Subhash,
>
> I hope you're ready to do some homework! :) When you say "control
> FreeSWITCH" that can mean different things:
> * Configuring FreeSWITCH
> * Watching FreeSWITCH to see what calls/activity are currently taking place
> * Generating new calls (outbound IVR, e.g.)
> * Handling inbound calls (e.g. IVR/Auto Attendant)
>
> Could you offer some more details about what you are trying to do with
> FreeSWITCH? That will help us know whether or not ESL (event socket
> library) is the right thing to recommend.
>
> Thanks,
> Michael
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 krice at freeswitch.org  Tue Aug 28 08:14:13 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 27 Aug 2012 23:14:13 -0500
Subject: [Freeswitch-users] How to write an Application to control
	freeSWITCH
In-Reply-To: <CAFvodxW6435KvLQHUEkSiq+BB+BpE6fdF_mRv17eMy_AXEKzog@mail.gmail.com>
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
	<CALn233py3x9eJgXAvsE6q4dL85i4ho7cJU9aF9y5Z2sit1Q_TQ@mail.gmail.com>
	<CAFvodxWB5iBu0_SGbn4tyRn7h1Jyofcc5kA5Dk0mAimjkNKAnw@mail.gmail.com>
	<CAKzWOxVhM7Qv4LqOsTRWTUugjpAxeJoB6ep9SonfTHm-nNiPMA@mail.gmail.com>
	<CAFvodxW6435KvLQHUEkSiq+BB+BpE6fdF_mRv17eMy_AXEKzog@mail.gmail.com>
Message-ID: <880E607C-7F98-49FA-9BFD-74CD429CEF44@freeswitch.org>

step one get both books the bridge book and the cookbook
step two read them both cover to cover
step three try to implement the things the books are talking about
step four read the wiki for additional information



Ken
Sent from my iPad

On Aug 27, 2012, at 10:14 PM, Subhash <gvvsubhashkumar at gmail.com> wrote:

> Michael,
> 
>            You are right we are more concern about the below things but i would like to add more things in your list. Please find my comments in red.
> 
>     * Configuring FreeSWITCH(this include all conf files and dialplan generation ??)
>     * Watching FreeSWITCH to see what calls/activity are currently taking place
>     * Generating new calls (outbound IVR, e.g.)
>     * Handling inbound calls (e.g. IVR/Auto Attendant).
>     * How to use freeswitch for billing purpose.
>     *Bridging and Transferring the inbound calls.
> 
> 
> 
> Thanks,
> Subhash.
> 
> 
> On Mon, Aug 27, 2012 at 11:46 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
> 
> On Mon, Aug 27, 2012 at 9:42 AM, Subhash <gvvsubhashkumar at gmail.com> wrote:
> Hi,
> 
> Thanks for the reply .if you dont mind can you point out the url to get there because I tried but did not get needed information.
> 
> Thanjs,
> Subhash.
> 
> 
> Subhash,
> 
> I hope you're ready to do some homework! :) When you say "control FreeSWITCH" that can mean different things:
> * Configuring FreeSWITCH
> * Watching FreeSWITCH to see what calls/activity are currently taking place
> * Generating new calls (outbound IVR, e.g.)
> * Handling inbound calls (e.g. IVR/Auto Attendant)
> 
> Could you offer some more details about what you are trying to do with FreeSWITCH? That will help us know whether or not ESL (event socket library) is the right thing to recommend.
> 
> Thanks,
> Michael
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lists at kavun.ch  Tue Aug 28 09:03:40 2012
From: lists at kavun.ch (Emrah)
Date: Tue, 28 Aug 2012 01:03:40 -0400
Subject: [Freeswitch-users] Attended transfer to a conference room
Message-ID: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>

Hi all,

I am experiencing a strange issue with SIP based attended transfers.

If I call a number via a gateway and attend-transfer it to a SIP phone, it works. If I do the same but transfer the call into a conference extension instead, the line that is being transfered is hanged up.
There is no much activity on the SIP side of things, it seems to be very much related to FS.

Some info:
I call out through a provider configured on the external profile, from a phone registered on the internal profile.
It is not a codec conflict.
Both lines are answered when I actually finalize the transfer.
I tried with multiple phones and softphones.

I can clean up my logs and post them here, but if you guys have some info already it would be much appreciated.

Best,
Emrah

From gabe at gundy.org  Tue Aug 28 09:10:33 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 27 Aug 2012 23:10:33 -0600
Subject: [Freeswitch-users] Calls drop after ~1:48
In-Reply-To: <BAY156-W267FC7C8D62A763110D51FD4A20@phx.gbl>
References: <BAY156-W267FC7C8D62A763110D51FD4A20@phx.gbl>
Message-ID: <CAMwZ-tJrgLRpmyxEVHJCdaTxyY_7kxkgWegegxp7Vw0dg03cEw@mail.gmail.com>

On Sun, Aug 26, 2012 at 11:10 PM, R W <wingcomm at hotmail.com> wrote:
> We fixed this by shutting of the "Passive Session Timer" in the "SIP
> Miscellaneous." However, I'm curious to know if anyone can explain why this
> happens on some NATs and not others? Is there s problem with disabling the
> "Passive Session Timer" on our clients?

I think you'll find it has less to do with NATing and more to do with
the SIP user agent (UA) behind the NAT. If it *does* have to do with
NATing, then it's likely the SIP 'helpers' getting in the way. Google
ALG-based SIP-capable firewalls.

Let us know what you find. Oh, and you'll want to check this at the
SIP message level to get it figured out. Good luck!


Best,
Gabe


From sharad at coraltele.com  Tue Aug 28 09:11:51 2012
From: sharad at coraltele.com (Sharad Garg)
Date: Tue, 28 Aug 2012 10:41:51 +0530
Subject: [Freeswitch-users] How to write an Application to
	controlfreeSWITCH
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
	<CALn233py3x9eJgXAvsE6q4dL85i4ho7cJU9aF9y5Z2sit1Q_TQ@mail.gmail.com>
Message-ID: <4BA9BAACFB75451B85C50AF41025A1DE@coralocs.com>

Hello Subhash

If you may elobrate little more, I think we will be able to help you out.

Just explain what you want to achieve.

Best Regards
Sharad



----- Original Message ----- 
From: "Mitch Capper" <mitch.capper at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, August 27, 2012 9:23 PM
Subject: Re: [Freeswitch-users] How to write an Application to 
controlfreeSWITCH


> look at the wiki for esl probably the easiest way if you need to you
> can also write fs modules in c++.
> ~mitch
>
> On Mon, Aug 27, 2012 at 8:19 AM, Subhash <gvvsubhashkumar at gmail.com> 
> wrote:
>>
>> Hi All,
>>
>>        I am new to freeSWITCH platform.
>>
>>       I want to write an application using c++ to control freeSWITCH so
>> please guide me.
>>
>>       Thanks in advance.
>>
>> Thanks,
>> Subhash.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 gvvsubhashkumar at gmail.com  Tue Aug 28 10:08:55 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Mon, 27 Aug 2012 23:08:55 -0700
Subject: [Freeswitch-users] How to write an Application to
	controlfreeSWITCH
In-Reply-To: <4BA9BAACFB75451B85C50AF41025A1DE@coralocs.com>
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
	<CALn233py3x9eJgXAvsE6q4dL85i4ho7cJU9aF9y5Z2sit1Q_TQ@mail.gmail.com>
	<4BA9BAACFB75451B85C50AF41025A1DE@coralocs.com>
Message-ID: <CAFvodxW8R5OCOC8YxF24dCdfzrPVHubCx4ApPFnai_wJH+OPRw@mail.gmail.com>

 Hi Sharad,

We are trying to achieve the follwing things from our app.

 Configuring FreeSWITCH
 Watching FreeSWITCH to see what calls/activity are currently taking place
 Call Bridging and transfer.
 Act as like an IVR.


Thanks,
Subhash.


On Mon, Aug 27, 2012 at 10:11 PM, Sharad Garg <sharad at coraltele.com> wrote:

> Hello Subhash
>
> If you may elobrate little more, I think we will be able to help you out.
>
> Just explain what you want to achieve.
>
> Best Regards
> Sharad
>
>
>
> ----- Original Message -----
> From: "Mitch Capper" <mitch.capper at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, August 27, 2012 9:23 PM
> Subject: Re: [Freeswitch-users] How to write an Application to
> controlfreeSWITCH
>
>
> > look at the wiki for esl probably the easiest way if you need to you
> > can also write fs modules in c++.
> > ~mitch
> >
> > On Mon, Aug 27, 2012 at 8:19 AM, Subhash <gvvsubhashkumar at gmail.com>
> > wrote:
> >>
> >> Hi All,
> >>
> >>        I am new to freeSWITCH platform.
> >>
> >>       I want to write an application using c++ to control freeSWITCH so
> >> please guide me.
> >>
> >>       Thanks in advance.
> >>
> >> Thanks,
> >> Subhash.
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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
> >>
> >
> >
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at endigotech.com  Tue Aug 28 10:40:24 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 28 Aug 2012 02:40:24 -0400
Subject: [Freeswitch-users] How to write an Application to
	controlfreeSWITCH
In-Reply-To: <CAFvodxW8R5OCOC8YxF24dCdfzrPVHubCx4ApPFnai_wJH+OPRw@mail.gmail.com>
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
	<CALn233py3x9eJgXAvsE6q4dL85i4ho7cJU9aF9y5Z2sit1Q_TQ@mail.gmail.com>
	<4BA9BAACFB75451B85C50AF41025A1DE@coralocs.com>
	<CAFvodxW8R5OCOC8YxF24dCdfzrPVHubCx4ApPFnai_wJH+OPRw@mail.gmail.com>
Message-ID: <CAKV-jFxzfmvJgT+qN6eWorTS-N+ASu7m3tcGwMpm-0a44L5YgQ@mail.gmail.com>

I could name a half dozen ways to do this that's way easier and doesn't
require c. what's the point of doing of using c?

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 28, 2012 2:11 AM, "Subhash" <gvvsubhashkumar at gmail.com> wrote:

>  Hi Sharad,
>
> We are trying to achieve the follwing things from our app.
>
>  Configuring FreeSWITCH
>  Watching FreeSWITCH to see what calls/activity are currently taking place
>  Call Bridging and transfer.
>  Act as like an IVR.
>
>
> Thanks,
> Subhash.
>
>
> On Mon, Aug 27, 2012 at 10:11 PM, Sharad Garg <sharad at coraltele.com>wrote:
>
>> Hello Subhash
>>
>> If you may elobrate little more, I think we will be able to help you out.
>>
>> Just explain what you want to achieve.
>>
>> Best Regards
>> Sharad
>>
>>
>>
>> ----- Original Message -----
>> From: "Mitch Capper" <mitch.capper at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, August 27, 2012 9:23 PM
>> Subject: Re: [Freeswitch-users] How to write an Application to
>> controlfreeSWITCH
>>
>>
>> > look at the wiki for esl probably the easiest way if you need to you
>> > can also write fs modules in c++.
>> > ~mitch
>> >
>> > On Mon, Aug 27, 2012 at 8:19 AM, Subhash <gvvsubhashkumar at gmail.com>
>> > wrote:
>> >>
>> >> Hi All,
>> >>
>> >>        I am new to freeSWITCH platform.
>> >>
>> >>       I want to write an application using c++ to control freeSWITCH so
>> >> please guide me.
>> >>
>> >>       Thanks in advance.
>> >>
>> >> Thanks,
>> >> Subhash.
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.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
>> >>
>> >
>> >
>> >
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sharad at coraltele.com  Tue Aug 28 11:04:47 2012
From: sharad at coraltele.com (Sharad Garg)
Date: Tue, 28 Aug 2012 12:34:47 +0530
Subject: [Freeswitch-users] How to write an Application
	tocontrolfreeSWITCH
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com><CALn233py3x9eJgXAvsE6q4dL85i4ho7cJU9aF9y5Z2sit1Q_TQ@mail.gmail.com><4BA9BAACFB75451B85C50AF41025A1DE@coralocs.com>
	<CAFvodxW8R5OCOC8YxF24dCdfzrPVHubCx4ApPFnai_wJH+OPRw@mail.gmail.com>
Message-ID: <6939CF2F4EC941D5A253C207A8F8CF2C@coralocs.com>

My comments are in red.

regards
Sharad
  ----- Original Message ----- 
  From: Subhash 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, August 28, 2012 11:38 AM
  Subject: Re: [Freeswitch-users] How to write an Application tocontrolfreeSWITCH


   Hi Sharad,

  We are trying to achieve the follwing things from our app.

   Configuring FreeSWITCH 
  >>>> Configuration of FS deponds upon what exactly your requirements are. Deponding upon that, desired config files will be configured. So it is little vast...not easy to comment on this.

   Watching FreeSWITCH to see what calls/activity are currently taking place

  >>>>>> There are FS CLI commands available, which can be given either on FS CLI or through telnet (if you want to monitor the FS from your own app). These commands will let you know the running calls / call durations / call originators / receivers / used codecs, etc, etc. I think this will be more than enough for you.

    Bridging and transfer.

  >>>>> Are you using FS as call server too ? if yes, just go through the wiki, you will get everything to bridge / transfer the calls.

   Act as like an IVR.
  >>>>> This is really something interesting. For use the FS as IVR server, FS really is a great and robust plateform. To provide more flexibility to your IVR, you can write your own Javascripts. When a call is answered by FS through dialpla, let the call handed over to JS. Now write JS to play the IVR. This is what we generally do.

  if you need some more help on this, give us the flow of your IVR, we will suggest the desired dialplan settings & sample JS.

  Thanks,
  Subhash.
  Regards
  Sharad


  On Mon, Aug 27, 2012 at 10:11 PM, Sharad Garg <sharad at coraltele.com> wrote:

    Hello Subhash

    If you may elobrate little more, I think we will be able to help you out.

    Just explain what you want to achieve.

    Best Regards
    Sharad




    ----- Original Message -----
    From: "Mitch Capper" <mitch.capper at gmail.com>
    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
    Sent: Monday, August 27, 2012 9:23 PM
    Subject: Re: [Freeswitch-users] How to write an Application to
    controlfreeSWITCH



    > look at the wiki for esl probably the easiest way if you need to you
    > can also write fs modules in c++.
    > ~mitch
    >
    > On Mon, Aug 27, 2012 at 8:19 AM, Subhash <gvvsubhashkumar at gmail.com>
    > wrote:
    >>
    >> Hi All,
    >>
    >>        I am new to freeSWITCH platform.
    >>
    >>       I want to write an application using c++ to control freeSWITCH so
    >> please guide me.
    >>
    >>       Thanks in advance.
    >>
    >> Thanks,
    >> Subhash.
    >>
    >> _________________________________________________________________________
    >> Professional FreeSWITCH Consulting Services:
    >> consulting at freeswitch.org
    >> http://www.freeswitchsolutions.com
    >>
    >> 
    >> 
    >>
    >> Official FreeSWITCH Sites
    >> http://www.freeswitch.org
    >> http://wiki.freeswitch.org
    >> http://www.cluecon.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
    >>
    >
    >
    >


    _________________________________________________________________________
    Professional FreeSWITCH Consulting Services:
    consulting at freeswitch.org
    http://www.freeswitchsolutions.com

    
    

    Official FreeSWITCH Sites
    http://www.freeswitch.org
    http://wiki.freeswitch.org
    http://www.cluecon.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 evgeniy at bestnet.kharkov.ua  Tue Aug 28 14:43:17 2012
From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan)
Date: Tue, 28 Aug 2012 13:43:17 +0300
Subject: [Freeswitch-users] mod_radius_cdr problem
Message-ID: <503CA0C5.2030309@bestnet.kharkov.ua>

Hi all,
I am currently running two billing modules: mod_radius_cdr and 
mod_nibblebill. I'm using mod_nibblebill only for hangup the call when 
user's balance is 0, for all other things i'm using radius.
My problem is that when mod_nibblebill transfer the call to hangup 
extension - radius accounting is failing.

My hangup extension:
<extension name="hangup">
  <condition field="destination_number" expression="^(hangup)$">
   <action application="hangup" data="NORMAL_CLEARING"/>
  </condition>
</extension>

RADIUS log: http://pastebin.com/yeGPJZH9

I think my problem is that Freeswitch-Dst changing: in start packet 
Freeswitch-Dst = "0675420288" and in the stop packet Freeswitch-Dst = 
"hangup". But i don't know how to fix it.
-- 
Evgeniy Movlyan,
BestNet Ltd.


From david.villasmil.work at gmail.com  Tue Aug 28 15:29:09 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 28 Aug 2012 13:29:09 +0200
Subject: [Freeswitch-users] How to write an Application
	tocontrolfreeSWITCH
In-Reply-To: <6939CF2F4EC941D5A253C207A8F8CF2C@coralocs.com>
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
	<CALn233py3x9eJgXAvsE6q4dL85i4ho7cJU9aF9y5Z2sit1Q_TQ@mail.gmail.com>
	<4BA9BAACFB75451B85C50AF41025A1DE@coralocs.com>
	<CAFvodxW8R5OCOC8YxF24dCdfzrPVHubCx4ApPFnai_wJH+OPRw@mail.gmail.com>
	<6939CF2F4EC941D5A253C207A8F8CF2C@coralocs.com>
Message-ID: <B4232C1F-C9F8-4188-B7C3-1DF6BC5A4511@gmail.com>

Hello,

I didt all that with ESL. To me there's nothing like it to do everything you want exactly as you want it. For the billing part, you will probably need to write your own stuff based on cdrs (maybe posted to a different server via curl_xml)

You indeed need to read the books and start playing with FS, don't expect to have a complete solution in 5 minutes, unless you buy something, you need to GET TO KNOW FS inside out.

Have fun! I know I did!

David

---

David Villasmil


On Aug 28, 2012, at 9:04, "Sharad Garg" <sharad at coraltele.com> wrote:

> My comments are in red.
>  
> regards
> Sharad
> ----- Original Message -----
> From: Subhash
> To: FreeSWITCH Users Help
> Sent: Tuesday, August 28, 2012 11:38 AM
> Subject: Re: [Freeswitch-users] How to write an Application tocontrolfreeSWITCH
> 
>  Hi Sharad,
>  
> We are trying to achieve the follwing things from our app.
>  
>  Configuring FreeSWITCH 
> >>>> Configuration of FS deponds upon what exactly your requirements are. Deponding upon that, desired config files will be configured. So it is little vast...not easy to comment on this.
> 
>  Watching FreeSWITCH to see what calls/activity are currently taking place
>  
> >>>>>> There are FS CLI commands available, which can be given either on FS CLI or through telnet (if you want to monitor the FS from your own app). These commands will let you know the running calls / call durations / call originators / receivers / used codecs, etc, etc. I think this will be more than enough for you.
>  
>   Bridging and transfer.
>  
> >>>>> Are you using FS as call server too ? if yes, just go through the wiki, you will get everything to bridge / transfer the calls.
>  
>  Act as like an IVR.
> >>>>> This is really something interesting. For use the FS as IVR server, FS really is a great and robust plateform. To provide more flexibility to your IVR, you can write your own Javascripts. When a call is answered by FS through dialpla, let    the call handed over to JS. Now write JS to play the IVR. This is what we generally do.
>  
> if you need some more help on this, give us the flow of your IVR, we will suggest the desired dialplan settings & sample JS.
> 
> Thanks,
> Subhash.
> Regards
> Sharad
> 
> On Mon, Aug 27, 2012 at 10:11 PM, Sharad Garg <sharad at coraltele.com> wrote:
> Hello Subhash
> 
> If you may elobrate little more, I think we will be able to help you out.
> 
> Just explain what you want to achieve.
> 
> Best Regards
> Sharad
> 
> 
> 
> ----- Original Message -----
> From: "Mitch Capper" <mitch.capper at gmail.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, August 27, 2012 9:23 PM
> Subject: Re: [Freeswitch-users] How to write an Application to
> controlfreeSWITCH
> 
> 
> > look at the wiki for esl probably the easiest way if you need to you
> > can also write fs modules in c++.
> > ~mitch
> >
> > On Mon, Aug 27, 2012 at 8:19 AM, Subhash <gvvsubhashkumar at gmail.com>
> > wrote:
> >>
> >> Hi All,
> >>
> >>        I am new to freeSWITCH platform.
> >>
> >>       I want to write an application using c++ to control freeSWITCH so
> >> please guide me.
> >>
> >>       Thanks in advance.
> >>
> >> Thanks,
> >> Subhash.
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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
> >>
> >
> >
> >
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ntomer at newgen.co.in  Tue Aug 28 15:53:35 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Tue, 28 Aug 2012 17:23:35 +0530
Subject: [Freeswitch-users] mod_CallCenter: Agents not being logged out
In-Reply-To: <503B735D.4070409@gmail.com>
References: <02ba01cd8456$110470d0$330d5270$@co.in>
	<503B735D.4070409@gmail.com>
Message-ID: <01b501cd8513$c27a9a20$476fce60$@co.in>

Hi,

Thanks, I was able to log out the agent from FreeSWITCH console. And I was
also able to define two extensions in dialpna.conf where agents can call to
login/logout.

But I am not able to do it from outside the system. I am not familiar with
Python, is that the only way or it can be done through some other way as
well?

Regards

Nitin

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vbvbrj
Sent: Monday, August 27, 2012 6:47 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_CallCenter: Agents not being logged out

> Now I am trying to set an agent's status with this command at the
> console: callcenter_config [agent set status 'Logged out'] [1000 at default]
>
> This command is executed alright, but when I execute "callcenter_config
> agent list", the status of 1000 at default is still shown as "available";
> why so?

Please see carefully how the command is written. It must be:
callcenter_config agent set status [agent name] 'Logged Out'

> Also please tell me what should be the syntax if I want to set status
> when agent dials a number. And is there a way to set this status from
> outside, like from a web page click?

On wiki 
http://wiki.freeswitch.org/wiki/Mod_callcenter#Script_to_announce_members_po
sition 
is an example for agent-login and agent-logout dialplan extension.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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

Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.



From eagle.antonio at gmail.com  Tue Aug 28 15:51:27 2012
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 28 Aug 2012 11:51:27 +0000
Subject: [Freeswitch-users] mod_CallCenter: Agents not being logged out
In-Reply-To: <01b501cd8513$c27a9a20$476fce60$@co.in>
References: <02ba01cd8456$110470d0$330d5270$@co.in>
	<503B735D.4070409@gmail.com>
	<01b501cd8513$c27a9a20$476fce60$@co.in>
Message-ID: <CAF+3j-v7xkK6iGjiZg9H1A4zhqSsa_7A9MTAzmaroMMfZMYQ8A@mail.gmail.com>

Hello Nitin.

Try this
http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC

Probably fits your use case

2012/8/28 Nitin Tomer <ntomer at newgen.co.in>

> Hi,
>
> Thanks, I was able to log out the agent from FreeSWITCH console. And I was
> also able to define two extensions in dialpna.conf where agents can call to
> login/logout.
>
> But I am not able to do it from outside the system. I am not familiar with
> Python, is that the only way or it can be done through some other way as
> well?
>
> Regards
>
> Nitin
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vbvbrj
> Sent: Monday, August 27, 2012 6:47 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] mod_CallCenter: Agents not being logged out
>
> > Now I am trying to set an agent's status with this command at the
> > console: callcenter_config [agent set status 'Logged out'] [1000 at default
> ]
> >
> > This command is executed alright, but when I execute "callcenter_config
> > agent list", the status of 1000 at default is still shown as "available";
> > why so?
>
> Please see carefully how the command is written. It must be:
> callcenter_config agent set status [agent name] 'Logged Out'
>
> > Also please tell me what should be the syntax if I want to set status
> > when agent dials a number. And is there a way to set this status from
> > outside, like from a web page click?
>
> On wiki
>
> http://wiki.freeswitch.org/wiki/Mod_callcenter#Script_to_announce_members_po
> sition
> is an example for agent-login and agent-logout dialplan extension.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
> Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 asilva at wirelessmundi.com  Tue Aug 28 16:39:00 2012
From: asilva at wirelessmundi.com (Antonio)
Date: Tue, 28 Aug 2012 14:39:00 +0200
Subject: [Freeswitch-users] incoming call hangup if remote SDP contains
	a=crypto
Message-ID: <1346157540.11174.82.camel@marces.madrid.commsmundi.com>

Hi all,

I'm having a problem to make calls, when the destination answer, the
calls is hangup with incompatible destination.

Debugging, i could see that is because fs drops the call, because it
doesn't support ZRTP.

So, is it normal that freeswitch hangs up the call when in the remote
party send in the sdp "a=crypto:1 ... " ?

Should it ignore and try to established the call using the supported
methods in the sdp? 
Like for video calls, if the remote send video codecs, if we do not have
them we don't hangup the call... just make a simple audio call...


For incoming calls there is no problem.


REMOTE SDP:

2012-08-28 10:37:56.068381 [DEBUG] sofia.c:6122 Remote SDP:
v=0
o=- 163879879 1 IN IP4 127.0.0.1
s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)
t=0 0
a=group:BUNDLE audio video
m=audio 57293 RTP/SAVPF 103 104 0 8 106 105 13 126
c=IN IP4 80.28.139.221
a=rtcp:57293 IN IP4 80.28.139.221
a=candidate:3460286729 1 udp 2130714367 192.168.3.1 56196 typ host
generation 0
a=candidate:3460286729 2 udp 2130714367 192.168.3.1 56196 typ host
generation 0
a=candidate:1004190935 1 udp 2130714367 192.168.10.25 55956 typ host
generation 0
a=candidate:1004190935 2 udp 2130714367 192.168.10.25 55956 typ host
generation 0
a=candidate:1734362844 1 udp 1912610559 80.28.139.221 57293 typ srflx
generation 0
a=candidate:1734362844 2 udp 1912610559 80.28.139.221 57293 typ srflx
generation 0
a=candidate:3130178147 1 udp 1912610559 88.9.167.79 59009 typ srflx
generation 0
a=candidate:3130178147 2 udp 1912610559 88.9.167.79 59009 typ srflx
generation 0
a=candidate:2159818233 1 tcp 1694506751 192.168.3.1 52952 typ host
generation 0
a=candidate:2159818233 2 tcp 1694506751 192.168.3.1 52952 typ host
generation 0
a=candidate:1968783399 1 tcp 1694506751 192.168.10.25 45540 typ host
generation 0
a=candidate:1968783399 2 tcp 1694506751 192.168.10.25 45540 typ host
generation 0
a=ice-ufrag:SdZ956Hj4vXnNOhU
a=ice-pwd:PQ9aJYFiLGiUfKZqqDXxQdtF
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:1j6B4cDFNA3Q99XoFujj3wYYex4ThJJNpOgbGifl 
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=ssrc:2321878499 cname:qlN/YrHlo6j9iIJY
a=ssrc:2321878499 mslabel:w4BkX3vCh3pGAdk7f2fqpUd6NT7JS2miWIhu
a=ssrc:2321878499 label:w4BkX3vCh3pGAdk7f2fqpUd6NT7JS2miWIhu00
m=video 57293 RTP/SAVPF 100 101 102
c=IN IP4 80.28.139.221
a=rtcp:57293 IN IP4 80.28.139.221
a=candidate:3460286729 1 udp 2130714367 192.168.3.1 56196 typ host
generation 0
a=candidate:3460286729 2 udp 2130714367 192.168.3.1 56196 typ host
generation 0
a=candidate:1004190935 1 udp 2130714367 192.168.10.25 55956 typ host
generation 0
a=candidate:1004190935 2 udp 2130714367 192.168.10.25 55956 typ host
generation 0
a=candidate:1734362844 1 udp 1912610559 80.28.139.221 57293 typ srflx
generation 0
a=candidate:1734362844 2 udp 1912610559 80.28.139.221 57293 typ srflx
generation 0
a=candidate:3130178147 1 udp 1912610559 88.9.167.79 59009 typ srflx
generation 0
a=candidate:3130178147 2 udp 1912610559 88.9.167.79 59009 typ srflx
generation 0
a=candidate:2159818233 1 tcp 1694506751 192.168.3.1 52952 typ host
generation 0
a=candidate:2159818233 2 tcp 1694506751 192.168.3.1 52952 typ host
generation 0
a=candidate:1968783399 1 tcp 1694506751 192.168.10.25 45540 typ host
generation 0
a=candidate:1968783399 2 tcp 1694506751 192.168.10.25 45540 typ host
generation 0
a=ice-ufrag:SdZ956Hj4vXnNOhU
a=ice-pwd:PQ9aJYFiLGiUfKZqqDXxQdtF
a=mid:video
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:1j6B4cDFNA3Q99XoFujj3wYYex4ThJJNpOgbGifl 
a=rtpmap:100 VP8/90000
a=rtpmap:101 red/90000
a=rtpmap:102 ulpfec/90000




log when it fails:

2012-08-28 10:38:02.068442 [NOTICE] sofia.c:6853 Channel
[sofia/192.168.10.1/sip:2102 at 192.168.10.30:5062] has been answered
2012-08-28 10:38:02.068442 [DEBUG] sofia_glue.c:3948 Looking for
zrtp-hash
2012-08-28 10:38:02.068442 [DEBUG] sofia_glue.c:3926 Deciding whether to
pass zrtp-hash between legs
2012-08-28 10:38:02.068442 [DEBUG] sofia_glue.c:3928
CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-08-28 10:38:02.068442 [ERR] sofia_glue.c:4837 a=crypto in RTP/AVP,
refer to rfc3711
2012-08-28 10:38:02.068442 [DEBUG] switch_core_session.c:772 Send signal
sofia/192.168.1.2_nat/2303 at test.commsmundi.com [BREAK]
2012-08-28 10:38:02.068442 [DEBUG] switch_channel.c:2924
(sofia/192.168.1.2_nat/2303 at test.commsmundi.com) Callstate Change
RINGING -> HANGUP
2012-08-28 10:38:02.068442 [NOTICE] switch_channel.c:3412 Hangup
sofia/192.168.1.2_nat/2303 at test.commsmundi.com [CS_EXECUTE]
[INCOMPATIBLE_DESTINATION]



-- 
Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com
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From ntomer at newgen.co.in  Tue Aug 28 17:36:37 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Tue, 28 Aug 2012 19:06:37 +0530
Subject: [Freeswitch-users] mod_CallCenter: Agents not being logged out
In-Reply-To: <CAF+3j-v7xkK6iGjiZg9H1A4zhqSsa_7A9MTAzmaroMMfZMYQ8A@mail.gmail.com>
References: <02ba01cd8456$110470d0$330d5270$@co.in>	<503B735D.4070409@gmail.com>	<01b501cd8513$c27a9a20$476fce60$@co.in>
	<CAF+3j-v7xkK6iGjiZg9H1A4zhqSsa_7A9MTAzmaroMMfZMYQ8A@mail.gmail.com>
Message-ID: <01df01cd8522$31b0e030$9512a090$@co.in>

Thanks a lot. It worked J

 

This mailing list has been a tremendous source of information and have
drastically cut down my learning cycle. I am really grateful for that.

 

Now one more question, I want a user to dial some information in the IVR
system, and then the agent to be able to access that information. How can I
achieve that?

 

Thanks and Regards

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Antonio
Teixeira
Sent: Tuesday, August 28, 2012 5:21 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_CallCenter: Agents not being logged out

 

Hello Nitin.

Try this
http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC

Probably fits your use case 

2012/8/28 Nitin Tomer <ntomer at newgen.co.in>

Hi,

Thanks, I was able to log out the agent from FreeSWITCH console. And I was
also able to define two extensions in dialpna.conf where agents can call to
login/logout.

But I am not able to do it from outside the system. I am not familiar with
Python, is that the only way or it can be done through some other way as
well?

Regards

Nitin

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vbvbrj
Sent: Monday, August 27, 2012 6:47 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_CallCenter: Agents not being logged out

> Now I am trying to set an agent's status with this command at the
> console: callcenter_config [agent set status 'Logged out'] [1000 at default]
>
> This command is executed alright, but when I execute "callcenter_config
> agent list", the status of 1000 at default is still shown as "available";
> why so?

Please see carefully how the command is written. It must be:
callcenter_config agent set status [agent name] 'Logged Out'

> Also please tell me what should be the syntax if I want to set status
> when agent dials a number. And is there a way to set this status from
> outside, like from a web page click?

On wiki
http://wiki.freeswitch.org/wiki/Mod_callcenter#Script_to_announce_members_po
sition
is an example for agent-login and agent-logout dialplan extension.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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

Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL.


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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

 


Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From eagle.antonio at gmail.com  Tue Aug 28 17:40:39 2012
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Tue, 28 Aug 2012 13:40:39 +0000
Subject: [Freeswitch-users] mod_CallCenter: Agents not being logged out
In-Reply-To: <01df01cd8522$31b0e030$9512a090$@co.in>
References: <02ba01cd8456$110470d0$330d5270$@co.in>
	<503B735D.4070409@gmail.com>
	<01b501cd8513$c27a9a20$476fce60$@co.in>
	<CAF+3j-v7xkK6iGjiZg9H1A4zhqSsa_7A9MTAzmaroMMfZMYQ8A@mail.gmail.com>
	<01df01cd8522$31b0e030$9512a090$@co.in>
Message-ID: <CAF+3j-sYtj-HeMzNM6UQwacFVFfZcWGaxC3xum6OVY-PAUeTdw@mail.gmail.com>

Well thats Harder .

You can use an External DB or  if you hack mod_callcenter maybe pass some
SIP "Extra" Headers

http://wiki.freeswitch.org/wiki/Strip_SIP_Headers

And process them on the softphone or something like that but probably just
make a script in your language of choice
http://wiki.freeswitch.org/wiki/PHP_ESL
And Use a DB To Store The Channel Vars Where the client keys are and
the call uuid
http://wiki.freeswitch.org/wiki/Session_uuid


2012/8/28 Nitin Tomer <ntomer at newgen.co.in>

>  Thanks a lot. It worked J****
>
> ** **
>
> This mailing list has been a tremendous source of information and have
> drastically cut down my learning cycle. I am really grateful for that.****
>
> ** **
>
> Now one more question, I want a user to dial some information in the IVR
> system, and then the agent to be able to access that information. How can I
> achieve that?****
>
> ** **
>
> Thanks and Regards****
>
> ** **
>
> Nitin****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Antonio
> Teixeira
> *Sent:* Tuesday, August 28, 2012 5:21 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mod_CallCenter: Agents not being logged
> out****
>
> ** **
>
> Hello Nitin.
>
> Try this
> http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC
>
> Probably fits your use case ****
>
> 2012/8/28 Nitin Tomer <ntomer at newgen.co.in>****
>
> Hi,
>
> Thanks, I was able to log out the agent from FreeSWITCH console. And I was
> also able to define two extensions in dialpna.conf where agents can call to
> login/logout.
>
> But I am not able to do it from outside the system. I am not familiar with
> Python, is that the only way or it can be done through some other way as
> well?
>
> Regards
>
> Nitin
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vbvbrj
> Sent: Monday, August 27, 2012 6:47 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] mod_CallCenter: Agents not being logged out
>
> > Now I am trying to set an agent's status with this command at the
> > console: callcenter_config [agent set status 'Logged out'] [1000 at default
> ]
> >
> > This command is executed alright, but when I execute "callcenter_config
> > agent list", the status of 1000 at default is still shown as "available";
> > why so?
>
> Please see carefully how the command is written. It must be:
> callcenter_config agent set status [agent name] 'Logged Out'
>
> > Also please tell me what should be the syntax if I want to set status
> > when agent dials a number. And is there a way to set this status from
> > outside, like from a web page click?
>
> On wiki
>
> http://wiki.freeswitch.org/wiki/Mod_callcenter#Script_to_announce_members_po
> sition
> is an example for agent-login and agent-logout dialplan extension.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From tahir at ictinnovations.com  Tue Aug 28 17:59:34 2012
From: tahir at ictinnovations.com (tahir almas)
Date: Tue, 28 Aug 2012 18:59:34 +0500
Subject: [Freeswitch-users] Open Source unified autodialer software
	released
In-Reply-To: <CAKbxfG8TLHVdAdMojh_F6wQmmoz=dBGqT_sG=yvGc9F=0jpzMA@mail.gmail.com>
References: <CAOL0t+KN+6SOgz3H_XBemB_vsun9QxCzD2pJH03bDSMo5gXA0A@mail.gmail.com>
	<CC613A87.42A38%krice@freeswitch.org>
	<CAKbxfG8TLHVdAdMojh_F6wQmmoz=dBGqT_sG=yvGc9F=0jpzMA@mail.gmail.com>
Message-ID: <CAOL0t+JM0RJKZUkK6OMaqn5g_hKa12t++LtGQX0YnJ+yczsTTA@mail.gmail.com>

Hi Anthony

Front end of both applications  was developed over Drupal CMS ( which is
licensed as GPLv2 OR Later version) using Plivo API's   therefore possible
options were to release the code in GPLv2 or  GPLv3 also in past  you
recommended to release the code  in GPL

LGPL or GPL are ok for your case, AGPL is a little scary.
>


I have no concern to release the code in any open source license but my
understanding that we can release Drupal based code only in GPLv2 OR GPLv3
and can not release the code in other open source license

Looking for your comments

Regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT




On Tue, Aug 28, 2012 at 2:25 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> yah gpl v3 means "free as long as you don't use it" don't associate FS
> with that kind of license scheme please.
>
>
> On Mon, Aug 27, 2012 at 2:53 PM, Ken Rice <krice at freeswitch.org> wrote:
> > Why GPLv3?? GPLv3 goes against the spirit of the FreeSWITCH project in
> > general
> >
> >
> >
> > On 8/27/12 2:43 PM, "tahir almas" <tahir at ictinnovations.com> wrote:
> >
> > Pleased to update that both  ICTDialer http://www.ictdialer.org and
> ICTFAX
> > http://www.ictfax.org are being offered as GPL v 3.0  software
> >
> > Regards
> > Tahir Almas
> >
> > Managing Partner
> > ICT Innovations
> > http://www.ictinnovations.com
> > Leveraging open source in ICT
> >
> >
> ****************************************************************************************************************
> > NOTICE OF CONFIDENTIALITY
> > This communication including any information transmitted with it is
> intended
> > only for the use of the addressees and is confidential and may be
> protected
> > by legal privilege . If you are not an intended recipient, be aware that
> any
> > disclosure, copying, distribution or use of this e-mail or any
> attachment is
> > prohibited. If you have received this e-mail in error, please notify us
> > immediately by returning it to the sender and delete this copy from your
> > system. Thank you for your cooperation.
> >
> >
> >
> >
> > On Wed, Jan 18, 2012 at 10:04 PM, tahir almas <tahir at ictinnovations.com>
> > wrote:
> >
> > Thanks for valuable suggestions , will consider these recommendations
> >
> > Regards
> > Tahir Almas
> >
> > Managing Partner
> > ICT Innovations
> > http://www.ictinnovations.com
> > Leveraging open source in ICT
> >
> >
> ****************************************************************************************************************
> > NOTICE OF CONFIDENTIALITY
> > This communication including any information transmitted with it is
> intended
> > only for the use of the addressees and is confidential and may be
> protected
> > by legal privilege . If you are not an intended recipient, be aware that
> any
> > disclosure, copying, distribution or use of this e-mail or any
> attachment is
> > prohibited. If you have received this e-mail in error, please notify us
> > immediately by returning it to the sender and delete this copy from your
> > system. Thank you for your cooperation.
> >
> >
> >
> >
> > On Tue, Jan 17, 2012 at 10:22 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >
> > LGPL or GPL are ok for your case, AGPL is a little scary.   It really
> > depends if you have intent to stop anyone from using it or not, if you
> don't
> > then just choose MIT or BSD which are both compat with GPL and less
> > restrictive.  Remember, GPL doesn't like BSD because it says you can do
> > anything you want (including sell it or distribute it) it's hard to
> > understand at first.  GPL wants code to be free like a birdy soaring in
> the
> > sky not free like it costs no money,  so they actually get mad at
> licenses
> > with no restrictions that are considered free as in contain no
> restrictions
> > so you have a large political battle over weather you want it to run
> free in
> > a field be free of rules or be free of charge........AGPL takes it a step
> > further and says if someone downloads your project and runs it on their
> > website as a service, that it will be violating the license unless they
> > provide the code to their entire infrastructure for everyone to look at.
>  I
> > don't know if that will go over too well for most business ppl.
> >
> >
> > In reality debating licenses over code written in plaintext scripting
> > languages makes me chuckle a bit, just the C snob in me I suppose.
> Really
> > if you choose to write code in stuff like that you may as well expect
> people
> > to do whatever they want with it.
> >
> > I don't even really like debating licenses at all, that's why we chose
> BSD
> > and MPL both the (free of restrictions) variety of license.  The only
> > obligation in MPL is to show changes to the guy who wrote the code you
> are
> > changing so he can see if it helps his own personal cause or not.
> >
> >
> > On Tue, Jan 17, 2012 at 12:35 AM, tahir almas <tahir at ictinnovations.com>
> > wrote:
> >
> > Realy thankful for your suggestion, ICTDialer is developed over Drupal
> 7.0
> > Licensed as GPL version 2 or later so we have to license ICTDialer as GPL
> > compatible license
> >
> > What open source License you recommend for ICTDialer ?
> >
> > Regards
> >
> > Tahir Almas
> >
> > Managing Partner
> > ICT Innovations
> > http://www.ictinnovations.com
> > Leveraging open source in ICT
> >
> >
> ****************************************************************************************************************
> > NOTICE OF CONFIDENTIALITY
> > This communication including any information transmitted with it is
> intended
> > only for the use of the addressees and is confidential and may be
> protected
> > by legal privilege . If you are not an intended recipient, be aware that
> any
> > disclosure, copying, distribution or use of this e-mail or any
> attachment is
> > prohibited. If you have received this e-mail in error, please notify us
> > immediately by returning it to the sender and delete this copy from your
> > system. Thank you for your cooperation.
> >
> >
> >
> >
> > On Tue, Jan 17, 2012 at 6:10 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >
> > One piece of advice is to not release it under AGPL which is the one that
> > triggers the copyleft over a socket and will turn away 99% of your
> > perspective testers.
> >
> >
> > On Mon, Jan 16, 2012 at 1:46 PM, tahir almas <tahir at ictinnovations.com>
> > wrote:
> >
> > Pleased to announce the release of open source Fax , SMS and Voice
> > broadcasting software solution ICTDialer http://www.ictdialer.orgdeveloped
> > over reknown  Drupal Conent Mnagment System and powerfull Plivo
> > Communication framework , Your contribution and suggestions are welcome
> >
> > Regards
> > Tahir Almas
> >
> > Managing Partner
> > ICT Innovations
> > http://www.ictinnovations.com
> > Leveraging open source in ICT
> >
> >
> ****************************************************************************************************************
> > NOTICE OF CONFIDENTIALITY
> > This communication including any information transmitted with it is
> intended
> > only for the use of the addressees and is confidential and may be
> protected
> > by legal privilege . If you are not an intended recipient, be aware that
> any
> > disclosure, copying, distribution or use of this e-mail or any
> attachment is
> > prohibited. If you have received this e-mail in error, please notify us
> > immediately by returning it to the sender and delete this copy from your
> > system. Thank you for your cooperation.
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
> >
> >
> >
> > --
> > Ken
> > http://www.FreeSWITCH.org
> > http://www.ClueCon.com
> > http://www.OSTAG.org
> > irc.freenode.net #freeswitch
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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From avi at avimarcus.net  Tue Aug 28 18:02:52 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 28 Aug 2012 17:02:52 +0300
Subject: [Freeswitch-users] Subscribe to sofia:: events?
In-Reply-To: <CALNZuw=z+oqK61e_Gb6q5eM3dGN_PJt9PdAgb8YGHrVAqcuKhw@mail.gmail.com>
References: <CALNZuw=z+oqK61e_Gb6q5eM3dGN_PJt9PdAgb8YGHrVAqcuKhw@mail.gmail.com>
Message-ID: <CALNZuwkuY-wUy7w0KgdKoeeDR7Ky34UkGppMO+5q3Dpf1-sPxg@mail.gmail.com>

Ideas...? Or should I open a jira on this?

-Avi


On Sat, Aug 25, 2012 at 11:36 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I seem to be unable to subscribe to sofia events.
> Subscribing to CUSTOM doesn't show them. Subscribing to sofia::register
> responds: "-ERR no keywords supplied"
> Only subscribing to ALL shows them (with type CUSTOM).
>
> Am I doing something wrong? Is this a bug?
>
> -Avi
>
>
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From peter.olsson at visionutveckling.se  Tue Aug 28 18:12:56 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 28 Aug 2012 14:12:56 +0000
Subject: [Freeswitch-users] Subscribe to sofia:: events?
Message-ID: <1FFF97C269757C458224B7C895F35F15150C9F@cantor.std.visionutv.se>

"CUSTOM sofia::register" should work

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Avi Marcus
Skickat: den 28 augusti 2012 16:03
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Subscribe to sofia:: events?

Ideas...? Or should I open a jira on this?

-Avi

On Sat, Aug 25, 2012 at 11:36 PM, Avi Marcus <avi at avimarcus.net<mailto:avi at avimarcus.net>> wrote:
I seem to be unable to subscribe to sofia events.
Subscribing to CUSTOM doesn't show them. Subscribing to sofia::register responds: "-ERR no keywords supplied"
Only subscribing to ALL shows them (with type CUSTOM).

Am I doing something wrong? Is this a bug?

-Avi


!DSPAM:503ccd2732761726646270!
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From cmrienzo at gmail.com  Tue Aug 28 18:20:58 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Tue, 28 Aug 2012 10:20:58 -0400
Subject: [Freeswitch-users] Subscribe to sofia:: events?
In-Reply-To: <CALNZuw=z+oqK61e_Gb6q5eM3dGN_PJt9PdAgb8YGHrVAqcuKhw@mail.gmail.com>
References: <CALNZuw=z+oqK61e_Gb6q5eM3dGN_PJt9PdAgb8YGHrVAqcuKhw@mail.gmail.com>
Message-ID: <CAD-qRkj+=4Yb-n5fWrk6b-5gB=HKAFMYXDfG1j3bQJeTCOAVgw@mail.gmail.com>

Escape character is '^]'.
Content-Type: auth/request

auth ClueCon

Content-Type: command/reply
Reply-Text: +OK accepted

event plain CUSTOM sofia::register

Content-Type: command/reply
Reply-Text: +OK event listener enabled plain



On Sat, Aug 25, 2012 at 4:36 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I seem to be unable to subscribe to sofia events.
> Subscribing to CUSTOM doesn't show them. Subscribing to sofia::register
> responds: "-ERR no keywords supplied"
> Only subscribing to ALL shows them (with type CUSTOM).
>
> Am I doing something wrong? Is this a bug?
>
> -Avi
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 avi at avimarcus.net  Tue Aug 28 18:33:08 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 28 Aug 2012 17:33:08 +0300
Subject: [Freeswitch-users] Subscribe to sofia:: events?
In-Reply-To: <CAD-qRkj+=4Yb-n5fWrk6b-5gB=HKAFMYXDfG1j3bQJeTCOAVgw@mail.gmail.com>
References: <CALNZuw=z+oqK61e_Gb6q5eM3dGN_PJt9PdAgb8YGHrVAqcuKhw@mail.gmail.com>
	<CAD-qRkj+=4Yb-n5fWrk6b-5gB=HKAFMYXDfG1j3bQJeTCOAVgw@mail.gmail.com>
Message-ID: <CALNZuwmgg0-NVs7aHb13J0uD65B54rF_E29jhxFCJNiX1fOQQg@mail.gmail.com>

That worked -- thanks!
-Avi Marcus

718-989-9485 (USA)
054-844-3271 (Israel Kosher)
077-228-5055 (Israel Landline)
020-3298-2875 (UK)



On Tue, Aug 28, 2012 at 5:20 PM, Christopher Rienzo <cmrienzo at gmail.com>wrote:

> Escape character is '^]'.
> Content-Type: auth/request
>
> auth ClueCon
>
> Content-Type: command/reply
> Reply-Text: +OK accepted
>
>  event plain CUSTOM sofia::register
>
> Content-Type: command/reply
> Reply-Text: +OK event listener enabled plain
>
>
>
> On Sat, Aug 25, 2012 at 4:36 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> I seem to be unable to subscribe to sofia events.
>> Subscribing to CUSTOM doesn't show them. Subscribing to sofia::register
>> responds: "-ERR no keywords supplied"
>> Only subscribing to ALL shows them (with type CUSTOM).
>>
>> Am I doing something wrong? Is this a bug?
>>
>> -Avi
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 babak.freeswitch at gmail.com  Tue Aug 28 18:41:29 2012
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Tue, 28 Aug 2012 19:11:29 +0430
Subject: [Freeswitch-users] directory variables not set on user channels
	when bridge does not succeed
Message-ID: <CAEVfcabHUn+Xy-3FSeYiWGcnPrV_6mubrf2VRavEqwtbHbgY=w@mail.gmail.com>

Hi
To add some features to fs, I need to set some variables in xml directory
user as:
<include>
  <user id="1000">
    <params>
     ...
    <variables>
  <variable name="profile_id" value="10"/>
  <variable name="dn_id" value="25"/>
  .....
  <variable name="call_timeout" value="7"/>
    </variables>
  </user>
</include>
as documentation says these variables are set on every inbound or outbound
channel for this user. when I try to call to user/1000 if call succeeds
everything is ok but if call ends with results like no_answer, in xml_cdr
that is sent to my cdr logger url, there is no sign of any variables set in
directory as above. is this normal? because if I set these on bleg
(user/1000) using export they appear in xml cdr.
thanx
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From msc at freeswitch.org  Tue Aug 28 18:53:33 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 28 Aug 2012 07:53:33 -0700
Subject: [Freeswitch-users] directory variables not set on user channels
 when bridge does not succeed
In-Reply-To: <CAEVfcabHUn+Xy-3FSeYiWGcnPrV_6mubrf2VRavEqwtbHbgY=w@mail.gmail.com>
References: <CAEVfcabHUn+Xy-3FSeYiWGcnPrV_6mubrf2VRavEqwtbHbgY=w@mail.gmail.com>
Message-ID: <CAKzWOxU+dg4kVvgOOxf6d7riWpNBWBoqJy-qy91WgBQu+PaDTA@mail.gmail.com>

If the bridge to the user is not successful then the channel variables
aren't set. If you need these variables no matter what then you can
probably use this:

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

Call that as soon as you know which user you're going to bridge to and it
should slurp in all the channel variables for that user even if the bridge
is not successful.

-MC

On Tue, Aug 28, 2012 at 7:41 AM, Babak Yakhchali <babak.freeswitch at gmail.com
> wrote:

> Hi
> To add some features to fs, I need to set some variables in xml directory
> user as:
> <include>
>   <user id="1000">
>     <params>
>      ...
>     <variables>
>   <variable name="profile_id" value="10"/>
>   <variable name="dn_id" value="25"/>
>    .....
>   <variable name="call_timeout" value="7"/>
>     </variables>
>   </user>
> </include>
> as documentation says these variables are set on every inbound or outbound
> channel for this user. when I try to call to user/1000 if call succeeds
> everything is ok but if call ends with results like no_answer, in xml_cdr
> that is sent to my cdr logger url, there is no sign of any variables set in
> directory as above. is this normal? because if I set these on bleg
> (user/1000) using export they appear in xml cdr.
> thanx
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Tue Aug 28 18:55:19 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 28 Aug 2012 07:55:19 -0700
Subject: [Freeswitch-users] voicemail to email not working
In-Reply-To: <503BF44F.6050007@gmail.com>
References: <503BF44F.6050007@gmail.com>
Message-ID: <CAKzWOxVDyQaAYqAEniOGmCGRzoszJZH_j0c30J-omUJA8=MHkQ@mail.gmail.com>

What MTA are you using? I know that I've had much more success using Tony's
sendmail replacement script. (src/scripts/perl/sendmail).

-MC

On Mon, Aug 27, 2012 at 3:27 PM, Abaci <abaci64 at gmail.com> wrote:

>  I'm trying to get FreeSWITCH to send my voicemail to my email and can't
> it to work. I have sSMTP installed and working (tested using voicemail.tpl)
> I see in the FreeSWITCH logs *"switch_utils.c:761 Emailed file
> [/tmp/mail.13461330724a96] to [me at mydomain.com]" *however in
> /var/log/maillog I see nothing and I don't get the email. does someone have
> a clue on what might be wrong here, or is there any debug I can enable for
> this?
>
> *in switch.conf.xml I have:*
>     <param name="mailer-app" value="sendmail"/>
>     <param name="mailer-app-args" value="-t"/>
>
> *in voicemail.conf.xml I have:*
>     <email>
>         <param name="template-file" value="voicemail.tpl"/>
>         <param name="notify-template-file" value="notify-voicemail.tpl"/>
>         <!-- this is the format voicemail_time will have -->
>         <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
>         <!--<param name="email-from" value="${voicemail_account}@
> ${voicemail_domain}"/>-->
>     </email>
>
> *in the directory (xml_curl) I have:*
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="mydomain.com">
>       <user id="200">
>         <params>
>           <param name="password" value="cj987r5y95"/>
>           <param name="vm-password" value="1234"/>
>           <param name="vm-enabled" value="true"/>
>           <param name="vm-email-all-messages" value="true"/>
>           <param name="vm-mailto" value="me at mydomain.com"<me at mydomain.com>
> />
>           <param name="vm-mailfrom" value="Voicemail Server
> <voicemail at mydomain.com> <voicemail at mydomain.com>"/>
>           <param name="vm-attach-file" value="true" />
>           <param name="dial-string" value=
> "${sofia_contact(200 at mydomain.com)}"<$%7Bsofia_contact(200 at mydomain.com)%7D>
> />
>         </params>
>         <variables>
>           <variable name="toll_allow"
> value="domestic,international,local"/>
>           <variable name="user_context" value="routing"/>
>           <variable name="effective_caller_id_name" value="test user"/>
>           <variable name="effective_caller_id_number" value="2125551212"/>
>           <variable name="outbound_caller_id_number" value="2125551212"/>
>         </variables>
>       </user>
>     </domain>
>   </section>
> </document>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Tue Aug 28 18:56:26 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 28 Aug 2012 07:56:26 -0700
Subject: [Freeswitch-users] Attended transfer to a conference room
In-Reply-To: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
References: <F7C50460-8361-4E57-AE26-58E96F34CEC7@kavun.ch>
Message-ID: <CAKzWOxXQ2mtimBNJRS4yntrEt1bW3qCsiErkip=uSxH8U_t9tw@mail.gmail.com>

Go ahead and clean up the logs and put them on pastebin.freeswitch.org.
-MC

On Mon, Aug 27, 2012 at 10:03 PM, Emrah <lists at kavun.ch> wrote:

> Hi all,
>
> I am experiencing a strange issue with SIP based attended transfers.
>
> If I call a number via a gateway and attend-transfer it to a SIP phone, it
> works. If I do the same but transfer the call into a conference extension
> instead, the line that is being transfered is hanged up.
> There is no much activity on the SIP side of things, it seems to be very
> much related to FS.
>
> Some info:
> I call out through a provider configured on the external profile, from a
> phone registered on the internal profile.
> It is not a codec conflict.
> Both lines are answered when I actually finalize the transfer.
> I tried with multiple phones and softphones.
>
> I can clean up my logs and post them here, but if you guys have some info
> already it would be much appreciated.
>
> Best,
> Emrah
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From abaci64 at gmail.com  Tue Aug 28 19:22:42 2012
From: abaci64 at gmail.com (Abaci)
Date: Tue, 28 Aug 2012 11:22:42 -0400
Subject: [Freeswitch-users] voicemail to email not working
In-Reply-To: <CAKzWOxVDyQaAYqAEniOGmCGRzoszJZH_j0c30J-omUJA8=MHkQ@mail.gmail.com>
References: <503BF44F.6050007@gmail.com>
	<CAKzWOxVDyQaAYqAEniOGmCGRzoszJZH_j0c30J-omUJA8=MHkQ@mail.gmail.com>
Message-ID: <503CE242.3000108@gmail.com>

using sSMTP 
<http://wiki.freeswitch.org/wiki/Mod_voicemail#sSMTP_settings>which also 
replaces sendmail

On 8/28/2012 10:55 AM, Michael Collins wrote:
> What MTA are you using? I know that I've had much more success using 
> Tony's sendmail replacement script. (src/scripts/perl/sendmail).
>
> -MC
>
> On Mon, Aug 27, 2012 at 3:27 PM, Abaci <abaci64 at gmail.com 
> <mailto:abaci64 at gmail.com>> wrote:
>
>     I'm trying to get FreeSWITCH to send my voicemail to my email and
>     can't it to work. I have sSMTP installed and working (tested using
>     voicemail.tpl) I see in the FreeSWITCH logs *"switch_utils.c:761
>     Emailed file [/tmp/mail.13461330724a96] to [me at mydomain.com
>     <mailto:me at mydomain.com>]" *however in /var/log/maillog I see
>     nothing and I don't get the email. does someone have a clue on
>     what might be wrong here, or is there any debug I can enable for this?
>
>     _*in switch.conf.xml I have:*_
>         <param name="mailer-app" value="sendmail"/>
>         <param name="mailer-app-args" value="-t"/>
>
>     _*in voicemail.conf.xml I have:*_
>         <email>
>             <param name="template-file" value="voicemail.tpl"/>
>             <param name="notify-template-file"
>     value="notify-voicemail.tpl"/>
>             <!-- this is the format voicemail_time will have -->
>             <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
>             <!--<param name="email-from"
>     value="${voicemail_account}@${voicemail_domain}"/>-->
>         </email>
>
>     _*in the directory (xml_curl) I have:*_
>     <document type="freeswitch/xml">
>       <section name="directory">
>         <domain name="mydomain.com <http://mydomain.com>">
>           <user id="200">
>             <params>
>               <param name="password" value="cj987r5y95"/>
>               <param name="vm-password" value="1234"/>
>               <param name="vm-enabled" value="true"/>
>               <param name="vm-email-all-messages" value="true"/>
>               <param name="vm-mailto" value="me at mydomain.com"
>     <mailto:me at mydomain.com>/>
>               <param name="vm-mailfrom" value="Voicemail Server
>     <voicemail at mydomain.com> <mailto:voicemail at mydomain.com>"/>
>               <param name="vm-attach-file" value="true" />
>               <param name="dial-string"
>     value="${sofia_contact(200 at mydomain.com)}"
>     <mailto:$%7Bsofia_contact%28200 at mydomain.com%29%7D>/>
>             </params>
>             <variables>
>               <variable name="toll_allow"
>     value="domestic,international,local"/>
>               <variable name="user_context" value="routing"/>
>               <variable name="effective_caller_id_name" value="test
>     user"/>
>               <variable name="effective_caller_id_number"
>     value="2125551212 <tel:2125551212>"/>
>               <variable name="outbound_caller_id_number"
>     value="2125551212 <tel:2125551212>"/>
>             </variables>
>           </user>
>         </domain>
>       </section>
>     </document>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     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
>
>
>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 babak.freeswitch at gmail.com  Tue Aug 28 19:28:20 2012
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Tue, 28 Aug 2012 19:58:20 +0430
Subject: [Freeswitch-users] directory variables not set on user channels
 when bridge does not succeed
In-Reply-To: <CAKzWOxU+dg4kVvgOOxf6d7riWpNBWBoqJy-qy91WgBQu+PaDTA@mail.gmail.com>
References: <CAEVfcabHUn+Xy-3FSeYiWGcnPrV_6mubrf2VRavEqwtbHbgY=w@mail.gmail.com>
	<CAKzWOxU+dg4kVvgOOxf6d7riWpNBWBoqJy-qy91WgBQu+PaDTA@mail.gmail.com>
Message-ID: <CAEVfcaY6vpYVoybC-vAB6H0a2ovvnsM2yWXpJt4UvvLu58NmBw@mail.gmail.com>

thanx
problem is I need to execute it on bleg, where the variables are not set. I
tried
session:setVariable('bridge_pre_execute_bleg_app','set_user')
session:setVariable('bridge_pre_execute_bleg_data',dn)
but it is not working until the b leg answers.
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From tnsampaio at bsd.com.br  Tue Aug 28 19:29:32 2012
From: tnsampaio at bsd.com.br (Tiago Sampaio)
Date: Tue, 28 Aug 2012 12:29:32 -0300
Subject: [Freeswitch-users] Ringing after queue exit
In-Reply-To: <503BBF06.8000306@gmail.com>
References: <503BB48B.1020404@bsd.com.br>
	<503BBF06.8000306@gmail.com>
Message-ID: <CAJxjjuv9yB7VSsnEYwHt1NyV5pG2zz=-0k2OzGHKY1vZJK7R3w@mail.gmail.com>

Thx, reading it i see ringback that solved my problem!



2012/8/27 Abaci <abaci64 at gmail.com>

> did you try setting transfer_rigback?
> http://wiki.freeswitch.org/wiki/Variable_transfer_ringback
> On 8/27/2012 1:55 PM, Tiago N. Sampaio wrote:
> > Hi all,
> >
> >       Im very new to FS, im comming from asterisk world hehe! Today i put
> > my 1st FS in operation!!!
> >
> >       Lets go, i have some queues in my company (queue1 queue2, etc..)
> >       I have max-wait on all queues to 60 seconds, and after that time i
> > wanna bridge caller with some user, ex:
> >
> >        <action application="callcenter" data="queue1 at default" />
> >        <action application="bridge" data="[leg_timeout=15]user/1002" />
> >        <action application="voicemail" data="default $${domain} 1002"/>
> >
> >       But when "max-wait-time" arrive, and fs try bridge to user/1002, no
> > ringing or moh playing, and callers think that call was droped..
> >       Only when voicemail is launched caller hear the sound of voicemail
> > announce...
> >
> >       Simplifying my question, how to put ringing on line when calling
> > user/1002 after unsucessfull join on queue1?
> >
> > Hugs
> > Tiago N. Sampaio
> >
> >
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Tiago N. Sampaio
BSD Certified Associate
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From daveh at beachdognet.com  Tue Aug 28 21:25:03 2012
From: daveh at beachdognet.com (Dave Horton)
Date: Tue, 28 Aug 2012 13:25:03 -0400
Subject: [Freeswitch-users] how to play a prompt after connection
In-Reply-To: <8B21E6D1-E702-460D-9FC8-B44449A8B99C@dchorton.com>
References: <8B21E6D1-E702-460D-9FC8-B44449A8B99C@dchorton.com>
Message-ID: <23F564DC-911A-4D5B-A63F-D420C755B216@beachdognet.com>


I need to do the following:

(1) connect an inbound caller to ivr, and collect a phone number
(2) bridge the caller to the entered phone number
(3) just after connection, play a prompt to the called party 'this call is being recorded'
(4) make a recording of the call

I basically have everything working except number 3 -- playing the prompt to the called party immediately after they pick up.  Since I am successfully recording the call I assume we have a conference going on freeswitch, and I should be able to play the prompt to the conference somehow (it's ok if the calling party hears it as well).

I'm doing this in lua, but I think if someone points me to the freeswitch commands to use to stream a file to both parties immediately after bridging that's all I need.





From abaci64 at gmail.com  Tue Aug 28 21:33:23 2012
From: abaci64 at gmail.com (Abaci)
Date: Tue, 28 Aug 2012 13:33:23 -0400
Subject: [Freeswitch-users] how to play a prompt after connection
In-Reply-To: <23F564DC-911A-4D5B-A63F-D420C755B216@beachdognet.com>
References: <8B21E6D1-E702-460D-9FC8-B44449A8B99C@dchorton.com>
	<23F564DC-911A-4D5B-A63F-D420C755B216@beachdognet.com>
Message-ID: <503D00E3.30800@gmail.com>

try api_on_answer or execute_on answer
http://wiki.freeswitch.org/wiki/Variable_api_on_answer
http://wiki.freeswitch.org/wiki/Variable_execute_on_answer

On 8/28/2012 1:25 PM, Dave Horton wrote:
> I need to do the following:
>
> (1) connect an inbound caller to ivr, and collect a phone number
> (2) bridge the caller to the entered phone number
> (3) just after connection, play a prompt to the called party 'this call is being recorded'
> (4) make a recording of the call
>
> I basically have everything working except number 3 -- playing the prompt to the called party immediately after they pick up.  Since I am successfully recording the call I assume we have a conference going on freeswitch, and I should be able to play the prompt to the conference somehow (it's ok if the calling party hears it as well).
>
> I'm doing this in lua, but I think if someone points me to the freeswitch commands to use to stream a file to both parties immediately after bridging that's all I need.
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 avi at avimarcus.net  Tue Aug 28 21:40:06 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 28 Aug 2012 20:40:06 +0300
Subject: [Freeswitch-users] how to play a prompt after connection
In-Reply-To: <503D00E3.30800@gmail.com>
References: <8B21E6D1-E702-460D-9FC8-B44449A8B99C@dchorton.com>
	<23F564DC-911A-4D5B-A63F-D420C755B216@beachdognet.com>
	<503D00E3.30800@gmail.com>
Message-ID: <CALNZuwkEpK77NNFe9jX1T8huBW4SLb5U1y4Xy=VOLNwrQ23aSQ@mail.gmail.com>

Set api_on_answer to execute
uuid_broadcast<http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast>
which
will let you pick which legs to play the recording to.
You can pass in the UUID with ${uuid}

-Avi


On Tue, Aug 28, 2012 at 8:33 PM, Abaci <abaci64 at gmail.com> wrote:

> try api_on_answer or execute_on answer
> http://wiki.freeswitch.org/wiki/Variable_api_on_answer
> http://wiki.freeswitch.org/wiki/Variable_execute_on_answer
>
> On 8/28/2012 1:25 PM, Dave Horton wrote:
> > I need to do the following:
> >
> > (1) connect an inbound caller to ivr, and collect a phone number
> > (2) bridge the caller to the entered phone number
> > (3) just after connection, play a prompt to the called party 'this call
> is being recorded'
> > (4) make a recording of the call
> >
> > I basically have everything working except number 3 -- playing the
> prompt to the called party immediately after they pick up.  Since I am
> successfully recording the call I assume we have a conference going on
> freeswitch, and I should be able to play the prompt to the conference
> somehow (it's ok if the calling party hears it as well).
> >
> > I'm doing this in lua, but I think if someone points me to the
> freeswitch commands to use to stream a file to both parties immediately
> after bridging that's all I need.
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Aug 28 22:05:20 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 28 Aug 2012 11:05:20 -0700
Subject: [Freeswitch-users] directory variables not set on user channels
 when bridge does not succeed
In-Reply-To: <CAEVfcaY6vpYVoybC-vAB6H0a2ovvnsM2yWXpJt4UvvLu58NmBw@mail.gmail.com>
References: <CAEVfcabHUn+Xy-3FSeYiWGcnPrV_6mubrf2VRavEqwtbHbgY=w@mail.gmail.com>
	<CAKzWOxU+dg4kVvgOOxf6d7riWpNBWBoqJy-qy91WgBQu+PaDTA@mail.gmail.com>
	<CAEVfcaY6vpYVoybC-vAB6H0a2ovvnsM2yWXpJt4UvvLu58NmBw@mail.gmail.com>
Message-ID: <CAKzWOxUST6r2=JB8r=2yxbUun=pU4vR=S+fbXRH7UEE=bREJ3Q@mail.gmail.com>

On Tue, Aug 28, 2012 at 8:28 AM, Babak Yakhchali <babak.freeswitch at gmail.com
> wrote:

> thanx
> problem is I need to execute it on bleg, where the variables are not set.
> I tried
> session:setVariable('bridge_pre_execute_bleg_app','set_user')
> session:setVariable('bridge_pre_execute_bleg_data',dn)
> but it is not working until the b leg answers.
>
>
Yeah, I can see the trouble with setting channel variables on a channel
that does not exist. Question: what happens when there is no b leg, i.e.,
when the called party does not answer? You don't have a b-leg CDR in that
case, do you?

-MC
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From msc at freeswitch.org  Tue Aug 28 22:05:51 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 28 Aug 2012 11:05:51 -0700
Subject: [Freeswitch-users] voicemail to email not working
In-Reply-To: <503CE242.3000108@gmail.com>
References: <503BF44F.6050007@gmail.com>
	<CAKzWOxVDyQaAYqAEniOGmCGRzoszJZH_j0c30J-omUJA8=MHkQ@mail.gmail.com>
	<503CE242.3000108@gmail.com>
Message-ID: <CAKzWOxWsAnW44y2OpF+CHB8O3eVMfJPgqsiGVTWin+Q8V-hb0w@mail.gmail.com>

and what, if anything, shows up in the sSMTP logs?

On Tue, Aug 28, 2012 at 8:22 AM, Abaci <abaci64 at gmail.com> wrote:

>  using sSMTP
> <http://wiki.freeswitch.org/wiki/Mod_voicemail#sSMTP_settings>which also
> replaces sendmail
>
> On 8/28/2012 10:55 AM, Michael Collins wrote:
>
> What MTA are you using? I know that I've had much more success using
> Tony's sendmail replacement script. (src/scripts/perl/sendmail).
>
> -MC
>
> On Mon, Aug 27, 2012 at 3:27 PM, Abaci <abaci64 at gmail.com> wrote:
>
>>  I'm trying to get FreeSWITCH to send my voicemail to my email and can't
>> it to work. I have sSMTP installed and working (tested using voicemail.tpl)
>> I see in the FreeSWITCH logs *"switch_utils.c:761 Emailed file
>> [/tmp/mail.13461330724a96] to [me at mydomain.com]" *however in
>> /var/log/maillog I see nothing and I don't get the email. does someone have
>> a clue on what might be wrong here, or is there any debug I can enable for
>> this?
>>
>> *in switch.conf.xml I have:*
>>     <param name="mailer-app" value="sendmail"/>
>>     <param name="mailer-app-args" value="-t"/>
>>
>> *in voicemail.conf.xml I have:*
>>     <email>
>>         <param name="template-file" value="voicemail.tpl"/>
>>         <param name="notify-template-file" value="notify-voicemail.tpl"/>
>>         <!-- this is the format voicemail_time will have -->
>>         <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
>>         <!--<param name="email-from" value="${voicemail_account}@
>> ${voicemail_domain}"/>-->
>>     </email>
>>
>> *in the directory (xml_curl) I have:*
>> <document type="freeswitch/xml">
>>   <section name="directory">
>>     <domain name="mydomain.com">
>>       <user id="200">
>>         <params>
>>           <param name="password" value="cj987r5y95"/>
>>           <param name="vm-password" value="1234"/>
>>           <param name="vm-enabled" value="true"/>
>>           <param name="vm-email-all-messages" value="true"/>
>>           <param name="vm-mailto" value="me at mydomain.com"<me at mydomain.com>
>> />
>>           <param name="vm-mailfrom" value="Voicemail Server
>> <voicemail at mydomain.com> <voicemail at mydomain.com>"/>
>>           <param name="vm-attach-file" value="true" />
>>           <param name="dial-string" value=
>> "${sofia_contact(200 at mydomain.com)}"<$%7Bsofia_contact%28200 at mydomain.com%29%7D>
>> />
>>         </params>
>>         <variables>
>>           <variable name="toll_allow"
>> value="domestic,international,local"/>
>>           <variable name="user_context" value="routing"/>
>>           <variable name="effective_caller_id_name" value="test user"/>
>>           <variable name="effective_caller_id_number" value="2125551212
>> "/>
>>           <variable name="outbound_caller_id_number" value="2125551212"/>
>>         </variables>
>>       </user>
>>     </domain>
>>   </section>
>> </document>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From w8hdkim at gmail.com  Tue Aug 28 22:36:23 2012
From: w8hdkim at gmail.com (Kim Culhan)
Date: Tue, 28 Aug 2012 14:36:23 -0400
Subject: [Freeswitch-users] voicemail to email not working
Message-ID: <CAKZxVQVHSpUju8c=rTh932HRPxKWFk_WrHK3=RRG3_+eUN5uSw@mail.gmail.com>

According to the docs sSMTP is usually used to send all messages to a
single address specified in the ssmtp.conf file
unless you set it up so 'address rewriting is disabled.'

http://linux.die.net/man/5/ssmtp.conf

Seems you might want to follow Michael's advice and use Tony's script
or use the original sendmail instead.

-kim

On Tue, August 28, 2012 11:22 am, Abaci wrote:
> using sSMTP
> <http://wiki.freeswitch.org/wiki/Mod_voicemail#sSMTP_settings>which also
> replaces sendmail
>
> On 8/28/2012 10:55 AM, Michael Collins wrote:
>> What MTA are you using? I know that I've had much more success using
>> Tony's sendmail replacement script. (src/scripts/perl/sendmail).
>>
>> -MC
>>
>> On Mon, Aug 27, 2012 at 3:27 PM, Abaci <abaci64 at gmail.com
>> <mailto:abaci64 at gmail.com>> wrote:
>>
>>     I'm trying to get FreeSWITCH to send my voicemail to my email and
>>     can't it to work. I have sSMTP installed and working (tested using
>>     voicemail.tpl) I see in the FreeSWITCH logs *"switch_utils.c:761
>>     Emailed file [/tmp/mail.13461330724a96] to [me at mydomain.com
>>     <mailto:me at mydomain.com>]" *however in /var/log/maillog I see
>>     nothing and I don't get the email. does someone have a clue on
>>     what might be wrong here, or is there any debug I can enable for this?
>>
>>     _*in switch.conf.xml I have:*_
>>         <param name="mailer-app" value="sendmail"/>
>>         <param name="mailer-app-args" value="-t"/>
>>
>>     _*in voicemail.conf.xml I have:*_
>>         <email>
>>             <param name="template-file" value="voicemail.tpl"/>
>>             <param name="notify-template-file"
>>     value="notify-voicemail.tpl"/>
>>             <!-- this is the format voicemail_time will have -->
>>             <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
>>             <!--<param name="email-from"
>>     value="${voicemail_account}@${voicemail_domain}"/>-->
>>         </email>
>>
>>     _*in the directory (xml_curl) I have:*_
>>     <document type="freeswitch/xml">
>>       <section name="directory">
>>         <domain name="mydomain.com <http://mydomain.com>">
>>           <user id="200">
>>             <params>
>>               <param name="password" value="cj987r5y95"/>
>>               <param name="vm-password" value="1234"/>
>>               <param name="vm-enabled" value="true"/>
>>               <param name="vm-email-all-messages" value="true"/>
>>               <param name="vm-mailto" value="me at mydomain.com"
>>     <mailto:me at mydomain.com>/>
>>               <param name="vm-mailfrom" value="Voicemail Server
>>     <voicemail at mydomain.com> <mailto:voicemail at mydomain.com>"/>
>>               <param name="vm-attach-file" value="true" />
>>               <param name="dial-string"
>>     value="${sofia_contact(200 at mydomain.com)}"
>>     <mailto:$%7Bsofia_contact%28200 at mydomain.com%29%7D>/>
>>             </params>
>>             <variables>
>>               <variable name="toll_allow"
>>     value="domestic,international,local"/>
>>               <variable name="user_context" value="routing"/>
>>               <variable name="effective_caller_id_name" value="test
>>     user"/>
>>               <variable name="effective_caller_id_number"
>>     value="2125551212 <tel:2125551212>"/>
>>               <variable name="outbound_caller_id_number"
>>     value="2125551212 <tel:2125551212>"/>
>>             </variables>
>>           </user>
>>         </domain>
>>       </section>
>>     </document>


From abaci64 at gmail.com  Tue Aug 28 22:37:48 2012
From: abaci64 at gmail.com (Abaci)
Date: Tue, 28 Aug 2012 14:37:48 -0400
Subject: [Freeswitch-users] voicemail to email not working
In-Reply-To: <CAKzWOxWsAnW44y2OpF+CHB8O3eVMfJPgqsiGVTWin+Q8V-hb0w@mail.gmail.com>
References: <503BF44F.6050007@gmail.com>
	<CAKzWOxVDyQaAYqAEniOGmCGRzoszJZH_j0c30J-omUJA8=MHkQ@mail.gmail.com>
	<503CE242.3000108@gmail.com>
	<CAKzWOxWsAnW44y2OpF+CHB8O3eVMfJPgqsiGVTWin+Q8V-hb0w@mail.gmail.com>
Message-ID: <503D0FFC.3000107@gmail.com>

that's the problem, I see nothing in the sSMTP logs (/var/log/maillog). 
I only see in the freeswitch logs *"switch_utils.c:761 Emailed file 
[/tmp/mail.13461330724a96] to [me at mydomain.com <mailto:me at mydomain.com>]"

*On 8/28/2012 2:05 PM, Michael Collins wrote:
> and what, if anything, shows up in the sSMTP logs?
>
> On Tue, Aug 28, 2012 at 8:22 AM, Abaci <abaci64 at gmail.com 
> <mailto:abaci64 at gmail.com>> wrote:
>
>     using sSMTP
>     <http://wiki.freeswitch.org/wiki/Mod_voicemail#sSMTP_settings>which also
>     replaces sendmail
>
>     On 8/28/2012 10:55 AM, Michael Collins wrote:
>>     What MTA are you using? I know that I've had much more success
>>     using Tony's sendmail replacement script.
>>     (src/scripts/perl/sendmail).
>>
>>     -MC
>>
>>     On Mon, Aug 27, 2012 at 3:27 PM, Abaci <abaci64 at gmail.com
>>     <mailto:abaci64 at gmail.com>> wrote:
>>
>>         I'm trying to get FreeSWITCH to send my voicemail to my email
>>         and can't it to work. I have sSMTP installed and working
>>         (tested using voicemail.tpl) I see in the FreeSWITCH logs
>>         *"switch_utils.c:761 Emailed file [/tmp/mail.13461330724a96]
>>         to [me at mydomain.com <mailto:me at mydomain.com>]" *however in
>>         /var/log/maillog I see nothing and I don't get the email.
>>         does someone have a clue on what might be wrong here, or is
>>         there any debug I can enable for this?
>>
>>         _*in switch.conf.xml I have:*_
>>             <param name="mailer-app" value="sendmail"/>
>>             <param name="mailer-app-args" value="-t"/>
>>
>>         _*in voicemail.conf.xml I have:*_
>>             <email>
>>                 <param name="template-file" value="voicemail.tpl"/>
>>                 <param name="notify-template-file"
>>         value="notify-voicemail.tpl"/>
>>                 <!-- this is the format voicemail_time will have -->
>>                 <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
>>                 <!--<param name="email-from"
>>         value="${voicemail_account}@${voicemail_domain}"/>-->
>>             </email>
>>
>>         _*in the directory (xml_curl) I have:*_
>>         <document type="freeswitch/xml">
>>           <section name="directory">
>>             <domain name="mydomain.com <http://mydomain.com>">
>>               <user id="200">
>>                 <params>
>>                   <param name="password" value="cj987r5y95"/>
>>                   <param name="vm-password" value="1234"/>
>>                   <param name="vm-enabled" value="true"/>
>>                   <param name="vm-email-all-messages" value="true"/>
>>                   <param name="vm-mailto" value="me at mydomain.com"
>>         <mailto:me at mydomain.com>/>
>>                   <param name="vm-mailfrom" value="Voicemail Server
>>         <voicemail at mydomain.com> <mailto:voicemail at mydomain.com>"/>
>>                   <param name="vm-attach-file" value="true" />
>>                   <param name="dial-string"
>>         value="${sofia_contact(200 at mydomain.com)}"
>>         <mailto:$%7Bsofia_contact%28200 at mydomain.com%29%7D>/>
>>                 </params>
>>                 <variables>
>>                   <variable name="toll_allow"
>>         value="domestic,international,local"/>
>>                   <variable name="user_context" value="routing"/>
>>                   <variable name="effective_caller_id_name"
>>         value="test user"/>
>>                   <variable name="effective_caller_id_number"
>>         value="2125551212 <tel:2125551212>"/>
>>                   <variable name="outbound_caller_id_number"
>>         value="2125551212 <tel:2125551212>"/>
>>                 </variables>
>>               </user>
>>             </domain>
>>           </section>
>>         </document>
>>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         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
>>
>>
>>
>>
>>     -- 
>>     Michael S Collins
>>     Twitter: @mercutioviz
>>     http://www.FreeSWITCH.org
>>     http://www.ClueCon.com
>>     http://www.OSTAG.org
>>
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     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
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     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
>
>
>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 abaci64 at gmail.com  Tue Aug 28 23:06:52 2012
From: abaci64 at gmail.com (Abaci)
Date: Tue, 28 Aug 2012 15:06:52 -0400
Subject: [Freeswitch-users] voicemail to email not working
In-Reply-To: <CAKzWOxWsAnW44y2OpF+CHB8O3eVMfJPgqsiGVTWin+Q8V-hb0w@mail.gmail.com>
References: <503BF44F.6050007@gmail.com>
	<CAKzWOxVDyQaAYqAEniOGmCGRzoszJZH_j0c30J-omUJA8=MHkQ@mail.gmail.com>
	<503CE242.3000108@gmail.com>
	<CAKzWOxWsAnW44y2OpF+CHB8O3eVMfJPgqsiGVTWin+Q8V-hb0w@mail.gmail.com>
Message-ID: <503D16CC.60502@gmail.com>

Tried anthony's perl sendmail script as the mailer-app and I get the 
following error:
*Can't call method "mail" on an undefined value at 
/usr/local/src/freeswitch/scripts/perl/sendmail line 36, <STDIN> chunk 1.
*
On 8/28/2012 2:05 PM, Michael Collins wrote:
> and what, if anything, shows up in the sSMTP logs?
>
> On Tue, Aug 28, 2012 at 8:22 AM, Abaci <abaci64 at gmail.com 
> <mailto:abaci64 at gmail.com>> wrote:
>
>     using sSMTP
>     <http://wiki.freeswitch.org/wiki/Mod_voicemail#sSMTP_settings>which also
>     replaces sendmail
>
>     On 8/28/2012 10:55 AM, Michael Collins wrote:
>>     What MTA are you using? I know that I've had much more success
>>     using Tony's sendmail replacement script.
>>     (src/scripts/perl/sendmail).
>>
>>     -MC
>>
>>     On Mon, Aug 27, 2012 at 3:27 PM, Abaci <abaci64 at gmail.com
>>     <mailto:abaci64 at gmail.com>> wrote:
>>
>>         I'm trying to get FreeSWITCH to send my voicemail to my email
>>         and can't it to work. I have sSMTP installed and working
>>         (tested using voicemail.tpl) I see in the FreeSWITCH logs
>>         *"switch_utils.c:761 Emailed file [/tmp/mail.13461330724a96]
>>         to [me at mydomain.com <mailto:me at mydomain.com>]" *however in
>>         /var/log/maillog I see nothing and I don't get the email.
>>         does someone have a clue on what might be wrong here, or is
>>         there any debug I can enable for this?
>>
>>         _*in switch.conf.xml I have:*_
>>             <param name="mailer-app" value="sendmail"/>
>>             <param name="mailer-app-args" value="-t"/>
>>
>>         _*in voicemail.conf.xml I have:*_
>>             <email>
>>                 <param name="template-file" value="voicemail.tpl"/>
>>                 <param name="notify-template-file"
>>         value="notify-voicemail.tpl"/>
>>                 <!-- this is the format voicemail_time will have -->
>>                 <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
>>                 <!--<param name="email-from"
>>         value="${voicemail_account}@${voicemail_domain}"/>-->
>>             </email>
>>
>>         _*in the directory (xml_curl) I have:*_
>>         <document type="freeswitch/xml">
>>           <section name="directory">
>>             <domain name="mydomain.com <http://mydomain.com>">
>>               <user id="200">
>>                 <params>
>>                   <param name="password" value="cj987r5y95"/>
>>                   <param name="vm-password" value="1234"/>
>>                   <param name="vm-enabled" value="true"/>
>>                   <param name="vm-email-all-messages" value="true"/>
>>                   <param name="vm-mailto" value="me at mydomain.com"
>>         <mailto:me at mydomain.com>/>
>>                   <param name="vm-mailfrom" value="Voicemail Server
>>         <voicemail at mydomain.com> <mailto:voicemail at mydomain.com>"/>
>>                   <param name="vm-attach-file" value="true" />
>>                   <param name="dial-string"
>>         value="${sofia_contact(200 at mydomain.com)}"
>>         <mailto:$%7Bsofia_contact%28200 at mydomain.com%29%7D>/>
>>                 </params>
>>                 <variables>
>>                   <variable name="toll_allow"
>>         value="domestic,international,local"/>
>>                   <variable name="user_context" value="routing"/>
>>                   <variable name="effective_caller_id_name"
>>         value="test user"/>
>>                   <variable name="effective_caller_id_number"
>>         value="2125551212 <tel:2125551212>"/>
>>                   <variable name="outbound_caller_id_number"
>>         value="2125551212 <tel:2125551212>"/>
>>                 </variables>
>>               </user>
>>             </domain>
>>           </section>
>>         </document>
>>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         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
>>
>>
>>
>>
>>     -- 
>>     Michael S Collins
>>     Twitter: @mercutioviz
>>     http://www.FreeSWITCH.org
>>     http://www.ClueCon.com
>>     http://www.OSTAG.org
>>
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     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
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     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
>
>
>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Tue Aug 28 23:54:35 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 28 Aug 2012 12:54:35 -0700
Subject: [Freeswitch-users] voicemail to email not working
In-Reply-To: <503D16CC.60502@gmail.com>
References: <503BF44F.6050007@gmail.com>
	<CAKzWOxVDyQaAYqAEniOGmCGRzoszJZH_j0c30J-omUJA8=MHkQ@mail.gmail.com>
	<503CE242.3000108@gmail.com>
	<CAKzWOxWsAnW44y2OpF+CHB8O3eVMfJPgqsiGVTWin+Q8V-hb0w@mail.gmail.com>
	<503D16CC.60502@gmail.com>
Message-ID: <CAKzWOxWFL-YG0nQHwC0crrBaYf_Z+A4RCS_sKcLQ48YQhp+Vpw@mail.gmail.com>

On Tue, Aug 28, 2012 at 12:06 PM, Abaci <abaci64 at gmail.com> wrote:

>  Tried anthony's perl sendmail script as the mailer-app and I get the
> following error:
> *Can't call method "mail" on an undefined value at
> /usr/local/src/freeswitch/scripts/perl/sendmail line 36, <STDIN> chunk 1.
> *
>
> Well, at least that's some feedback.
Line 36 appears to be this:
$smtp->mail($from);

I suspect the From: field is not being sent to the mail script because it's
not getting populated somewhere. You can try hard-coding the $from variable
for an ad-hoc test but I would definitely try to figure out why the field
is not being populated in the first place.

-MC
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From babak.freeswitch at gmail.com  Tue Aug 28 23:54:58 2012
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Wed, 29 Aug 2012 00:24:58 +0430
Subject: [Freeswitch-users] directory variables not set on user channels
 when bridge does not succeed
In-Reply-To: <CAKzWOxUST6r2=JB8r=2yxbUun=pU4vR=S+fbXRH7UEE=bREJ3Q@mail.gmail.com>
References: <CAEVfcabHUn+Xy-3FSeYiWGcnPrV_6mubrf2VRavEqwtbHbgY=w@mail.gmail.com>
	<CAKzWOxU+dg4kVvgOOxf6d7riWpNBWBoqJy-qy91WgBQu+PaDTA@mail.gmail.com>
	<CAEVfcaY6vpYVoybC-vAB6H0a2ovvnsM2yWXpJt4UvvLu58NmBw@mail.gmail.com>
	<CAKzWOxUST6r2=JB8r=2yxbUun=pU4vR=S+fbXRH7UEE=bREJ3Q@mail.gmail.com>
Message-ID: <CAEVfcabcAyrGvCFhVp_CX2f5fT-Zyqi6nqKH0vshL8nMLsinZw@mail.gmail.com>

there is a cdr. but user variables are not in it!
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From abaci64 at gmail.com  Tue Aug 28 23:59:05 2012
From: abaci64 at gmail.com (Abaci)
Date: Tue, 28 Aug 2012 15:59:05 -0400
Subject: [Freeswitch-users] voicemail to email not working
In-Reply-To: <CAKzWOxWFL-YG0nQHwC0crrBaYf_Z+A4RCS_sKcLQ48YQhp+Vpw@mail.gmail.com>
References: <503BF44F.6050007@gmail.com>
	<CAKzWOxVDyQaAYqAEniOGmCGRzoszJZH_j0c30J-omUJA8=MHkQ@mail.gmail.com>
	<503CE242.3000108@gmail.com>
	<CAKzWOxWsAnW44y2OpF+CHB8O3eVMfJPgqsiGVTWin+Q8V-hb0w@mail.gmail.com>
	<503D16CC.60502@gmail.com>
	<CAKzWOxWFL-YG0nQHwC0crrBaYf_Z+A4RCS_sKcLQ48YQhp+Vpw@mail.gmail.com>
Message-ID: <503D2309.30908@gmail.com>

I see that I only get feedback if I'm in the freeswitch console not on 
fs_cli, and yes the from was not populated. now that I get feedback I 
see that sSMTP doesn't like the "-t" so I took it off and things are 
working.
Thanks Michael and Kim

On 8/28/2012 3:54 PM, Michael Collins wrote:
>
>
> On Tue, Aug 28, 2012 at 12:06 PM, Abaci <abaci64 at gmail.com 
> <mailto:abaci64 at gmail.com>> wrote:
>
>     Tried anthony's perl sendmail script as the mailer-app and I get
>     the following error:
>     *Can't call method "mail" on an undefined value at
>     /usr/local/src/freeswitch/scripts/perl/sendmail line 36, <STDIN>
>     chunk 1.
>     *
>
> Well, at least that's some feedback.
> Line 36 appears to be this:
> $smtp->mail($from);
>
> I suspect the From: field is not being sent to the mail script because 
> it's not getting populated somewhere. You can try hard-coding the 
> $from variable for an ad-hoc test but I would definitely try to figure 
> out why the field is not being populated in the first place.
>
> -MC
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 Aug 29 00:14:01 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 28 Aug 2012 13:14:01 -0700
Subject: [Freeswitch-users] directory variables not set on user channels
 when bridge does not succeed
In-Reply-To: <CAEVfcabcAyrGvCFhVp_CX2f5fT-Zyqi6nqKH0vshL8nMLsinZw@mail.gmail.com>
References: <CAEVfcabHUn+Xy-3FSeYiWGcnPrV_6mubrf2VRavEqwtbHbgY=w@mail.gmail.com>
	<CAKzWOxU+dg4kVvgOOxf6d7riWpNBWBoqJy-qy91WgBQu+PaDTA@mail.gmail.com>
	<CAEVfcaY6vpYVoybC-vAB6H0a2ovvnsM2yWXpJt4UvvLu58NmBw@mail.gmail.com>
	<CAKzWOxUST6r2=JB8r=2yxbUun=pU4vR=S+fbXRH7UEE=bREJ3Q@mail.gmail.com>
	<CAEVfcabcAyrGvCFhVp_CX2f5fT-Zyqi6nqKH0vshL8nMLsinZw@mail.gmail.com>
Message-ID: <CAKzWOxUibiTNAHvv6K_=D-QcUDtq-3h3pjd_NYCg8wQrR-eRAw@mail.gmail.com>

That's because the user never actually answered. If you simply MUST have
these channel variables available on the b-leg, even if the b-leg does not
answer, then you'll need to get creative. Perhaps you can read them in and
export them in your bridge statement. Or you can post-process the CDR if
you know the user name. Sorry, it's the best you're gonna get with the
scenario you're dealing with.

-MC

On Tue, Aug 28, 2012 at 12:54 PM, Babak Yakhchali <
babak.freeswitch at gmail.com> wrote:

> there is a cdr. but user variables are not in it!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From babak.freeswitch at gmail.com  Wed Aug 29 00:44:00 2012
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Wed, 29 Aug 2012 01:14:00 +0430
Subject: [Freeswitch-users] directory variables not set on user channels
 when bridge does not succeed
In-Reply-To: <CAKzWOxUibiTNAHvv6K_=D-QcUDtq-3h3pjd_NYCg8wQrR-eRAw@mail.gmail.com>
References: <CAEVfcabHUn+Xy-3FSeYiWGcnPrV_6mubrf2VRavEqwtbHbgY=w@mail.gmail.com>
	<CAKzWOxU+dg4kVvgOOxf6d7riWpNBWBoqJy-qy91WgBQu+PaDTA@mail.gmail.com>
	<CAEVfcaY6vpYVoybC-vAB6H0a2ovvnsM2yWXpJt4UvvLu58NmBw@mail.gmail.com>
	<CAKzWOxUST6r2=JB8r=2yxbUun=pU4vR=S+fbXRH7UEE=bREJ3Q@mail.gmail.com>
	<CAEVfcabcAyrGvCFhVp_CX2f5fT-Zyqi6nqKH0vshL8nMLsinZw@mail.gmail.com>
	<CAKzWOxUibiTNAHvv6K_=D-QcUDtq-3h3pjd_NYCg8wQrR-eRAw@mail.gmail.com>
Message-ID: <CAEVfcaYnpiDWes=z6bnh=RR+XbvAM1Sjc1-gw_war64kpyfLZg@mail.gmail.com>

thanx.
 you are right, I get the info I need using username and dialed_user
channel variables
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From chris at gonumina.com  Wed Aug 29 01:00:12 2012
From: chris at gonumina.com (Chris Ferreira)
Date: Tue, 28 Aug 2012 17:00:12 -0400
Subject: [Freeswitch-users] Setup Voicemail to Expire after X amount of Days.
Message-ID: <CAEuuoePFxh2tK4hzxZV7Z1OF-ZNDS8FmJCVKpkFZiy0ndf25KA@mail.gmail.com>

How can I set FreeSWITCH up so that OLD voicemails get deleted
automatically?

I havent found anything in the Wiki or other documentation.



Your input is much appreciated.




Thanks,

-Chris
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From avi at avimarcus.net  Wed Aug 29 01:04:46 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 29 Aug 2012 00:04:46 +0300
Subject: [Freeswitch-users] Setup Voicemail to Expire after X amount of
	Days.
In-Reply-To: <CAEuuoePFxh2tK4hzxZV7Z1OF-ZNDS8FmJCVKpkFZiy0ndf25KA@mail.gmail.com>
References: <CAEuuoePFxh2tK4hzxZV7Z1OF-ZNDS8FmJCVKpkFZiy0ndf25KA@mail.gmail.com>
Message-ID: <CALNZuw=6mxvXdjE7cmVDhKUrT5SKZ82HO5TR5TpQyHLwbXnrcw@mail.gmail.com>

I don't think it exists. You could either custom-code a C module that polls
at various times..

Or just set a cron that queries the DB for old VMs then deletes the entry
and the file.

-Avi



On Wed, Aug 29, 2012 at 12:00 AM, Chris Ferreira <chris at gonumina.com> wrote:

> How can I set FreeSWITCH up so that OLD voicemails get deleted
> automatically?
>
> I havent found anything in the Wiki or other documentation.
>
>
>
> Your input is much appreciated.
>
>
>
>
> Thanks,
>
> -Chris
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 marketing at cluecon.com  Wed Aug 29 01:07:35 2012
From: marketing at cluecon.com (Michael Collins)
Date: Tue, 28 Aug 2012 14:07:35 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxVtqAVedgWH=H0XEpW+0bZ+0KiaNKhUPOzi-ZQfyjETmw@mail.gmail.com>

Hello all!

Apologies for the delay on this week's news and notes. Yesterday was quite
busy, but things are going well. I am happy to report that we have
submitted the first chapter of the new FreeSWITCH book to the publisher!
We've also conferred with a few members of the community and convinced
Packt<http://www.packtpub.com>to let us add some bonus content! Stay
tuned for more previews and teases.
Right now it's still early in the game so I don't want to reveal too much.

On last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22>we discussed
a number of things. First, we did a follow up to Ken's
previous discussion
<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_15>about the stable
1.2 branch and using git. Second, we talked a bit about
Vestec <http://www.vestec.com> and the great ASR application contest. (More
information is forthcoming!) Lastly, we had Mitch Capper discussing the
latest version of the FSClient
<http://wiki.freeswitch.org/wiki/FSClient>Windows softphone. If you
haven't tried it out I highly recommend it. It's
now stable and feature-enabled to the point that I've discontinued using
X-Lite or Jitsi.

Tomorrow we hope to have a discussion about
TLS<http://wiki.freeswitch.org/wiki/Tls>.
We have several community members who are experienced with key and
certificate management and we will be calling upon them to share their
experience with the rest of us. After that we will have an open discussion.

Cheers!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From daniel at pocock.com.au  Wed Aug 29 02:24:22 2012
From: daniel at pocock.com.au (Daniel Pocock)
Date: Wed, 29 Aug 2012 00:24:22 +0200
Subject: [Freeswitch-users] TLS key management (was FreeSWITCH Weekly
 News and Notes)
In-Reply-To: <CAKzWOxVtqAVedgWH=H0XEpW+0bZ+0KiaNKhUPOzi-ZQfyjETmw@mail.gmail.com>
References: <CAKzWOxVtqAVedgWH=H0XEpW+0bZ+0KiaNKhUPOzi-ZQfyjETmw@mail.gmail.com>
Message-ID: <503D4516.2060206@pocock.com.au>

On 28/08/12 23:07, Michael Collins wrote:

> Tomorrow we hope to have a discussion about
> TLS<http://wiki.freeswitch.org/wiki/Tls>.
> We have several community members who are experienced with key and
> certificate management and we will be calling upon them to share their
> experience with the rest of us. After that we will have an open discussion.


FYI, this came up at DebConf, some of us are thinking about similar
issues at a platform level:


http://penta.debconf.org/dc12_schedule/events/895.en.html

and notes from the session can be accessed with the gobby tool (sorry I
can't find this online somewhere, I've cut and pasted below for
convenience):

http://wiki.debian.org/gobby.debian.org

  look under debconf12/bof/X.509


As was emphasized at DebConf, it is a particular issue for VoIP
services, because different applications (e.g. a Jabber server and a SIP
server) can share the same key material (no need to waste money buying a
cert for each protocol), but sharing private keys between different
applications requires a more rigorous platform-wide effort to protect
the keys.






X.509 Certificate, Root CA, and Key best practices
==================================================

Confusing for packagers, admins, and users.  Let's fix!

Servers:
--------

 * where to store secret key material?  (/etc/ssl/private)
 * sharing key material across services?
 * locations for server certificates?
 * auto-creation of key/cert/certreq?
 * intermediate CA certificate storage
 * list of trusted root authorities for client certs -- shared?

Clients:
--------

 * list of trusted root authorities -- system store?  end-user override?
   -> Make all clients use the same store?
 * managing per-user certificates and secret keys -- p11-kit
 * how do we deal with root authorities for different contexts (web,
e-mail, code-signing, etc)
 * CRL/OCSP check?
   -> blacklist
   IIRC Microsoft now distributes blacklists via their update service
(chrome iirc too)
 * What can the cert be used for (email, website, software)
   -> mozilla has that information in it's store
   -> openssl can store it, but then uses a different header that others
might not be able to read

CSR process:
------------

(clients and servers) -- how can we make this less painful?

affected packages:
------------------

 * iceweasel/icedove (via nss and libckbi.so)
 * ca-certificates (potential split into ca-certificates,
ca-certificates-mozilla,
   ca-certificates-debconf, etc.)
 * ssl-cert
 * ca-certificates-java
 * kdelibs5-data: /usr/share/kde4/apps/kssl/ca-bundle.crt
 * nmap: /usr/share/ncat/ca-bundle.crt
 * ruby: /usr/lib/ruby/XXX/rubygems/ssl_certs/ca-bundle.pem
 * sympa: /usr/share/sympa/default/ca-bundle.crt
 * nutsurf-common: /usr/share/netsurf/ca-bundle.txt
 * abiword: /usr/share/abiword-2.9/certs/cacert.pem
 * acgvision-agent: /etc/acgvision/acgTrustStore.certs
 * apparmor: /etc/apparmor.d/abstractions/ssl_certs
 * gnupg2: /usr/share/gnupg2/com-certs.pem
 * libpurple0: /usr/share/purple/ca-certs/
 * linphone-common: /usr/share/linphone/rootca.pem
 * ntop-data: /usr/share/ntop/ntop-cert.pem
 * phoronix-test-suite
 * pidgin-microblog
 * burp
 * cpushare
 * acl2-books-certs
 * postgresql
 * openssl (embeds /etc/ssl/certs)
 * gnutls (no default authorities -- is this a good thing)
 * dovecot (and other IMAPs, POP3s servers)
 * exim4 and other MTAs (SMTPS, SMTP AUTH)
 * apache
 * vsftpd
 * resiprocate
 * XMPP servers
 * ...

normalizing and de-confusing names of directories:
--------------------------------------------------

 * /etc/ssl ? /etc/x509 ?
 * .../certs ? .../cas ?
 * /etc/ssl/public for non-ca certificates?

toolsmithing suggestions
------------------------

 * monitoring/review of sensitive directories (e.g. cronjob to review
these directories and report on changes,
   like many people do already with nagios/icinga)
 * tool to inspect/interrogate all open sockets that appear to support
TLS/SSL and report problems
 * something like db-config-common for X.509
 * ssl-cert and ca-certificates ca-certificates-java
 * per-machine root-authority that could be enabled by default?

Next steps:
-----------

 * best practices documentation: http://wiki.debian.org/X.509
 * what of this can or should be policy?
 * how can we get project-wide rough consensus
 * come up with a plan for how to file bugs; i.e. in the bug report,
   we need a suggested plan for what to do differently


Objectives:
-----------

 * make it easy for an administrator to configure his or her system for
X.509 certificates with a single easily auditable point of control
 * make it easy for a packager to rely on certificate creation,
management, expiry, etc. so that each package doesn't need to
reimplement these functions
 * make it easy for a user to use client-side certificates across tools
 * make it easy for a user to connect to public services predictably
with different tools

Other Distros Approaches
------------------------
 * Fedora's attempt to replace NSS
   http://fedoraproject.org/wiki/Nss_compat_ossl
   http://fedoraproject.org/wiki/FedoraCryptoConsolidation








From msc at freeswitch.org  Wed Aug 29 03:54:14 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 28 Aug 2012 16:54:14 -0700
Subject: [Freeswitch-users] Setup Voicemail to Expire after X amount of
	Days.
In-Reply-To: <CALNZuw=6mxvXdjE7cmVDhKUrT5SKZ82HO5TR5TpQyHLwbXnrcw@mail.gmail.com>
References: <CAEuuoePFxh2tK4hzxZV7Z1OF-ZNDS8FmJCVKpkFZiy0ndf25KA@mail.gmail.com>
	<CALNZuw=6mxvXdjE7cmVDhKUrT5SKZ82HO5TR5TpQyHLwbXnrcw@mail.gmail.com>
Message-ID: <CAKzWOxX2xf3uW_pXenPYa1tqTBr7qrDp0pa9aRB6+E9KkXE=DQ@mail.gmail.com>

Cron job is probably the lowest barrier to entry. Just get the sqlite3
client installed and go from there. Check out
fs-install/db/voicemail_default.db, specifically the voicemail_msgs table:

sqlite> .schema voicemail_msgs
CREATE TABLE voicemail_msgs (
   created_epoch INTEGER,
   read_epoch    INTEGER,
   username      VARCHAR(255),
   domain        VARCHAR(255),
   uuid          VARCHAR(255),
   cid_name      VARCHAR(255),
   cid_number    VARCHAR(255),
   in_folder     VARCHAR(255),
   file_path     VARCHAR(255),
   message_len   INTEGER,
   flags         VARCHAR(255),
   read_flags    VARCHAR(255),
   forwarded_by  VARCHAR(255)
);
CREATE INDEX voicemail_msgs_idx1 on voicemail_msgs(created_epoch);
CREATE INDEX voicemail_msgs_idx2 on voicemail_msgs(username);
CREATE INDEX voicemail_msgs_idx3 on voicemail_msgs(domain);
CREATE INDEX voicemail_msgs_idx4 on voicemail_msgs(uuid);
CREATE INDEX voicemail_msgs_idx5 on voicemail_msgs(in_folder);
CREATE INDEX voicemail_msgs_idx6 on voicemail_msgs(read_flags);
CREATE INDEX voicemail_msgs_idx7 on voicemail_msgs(forwarded_by);
CREATE INDEX voicemail_msgs_idx8 on voicemail_msgs(read_epoch);
CREATE INDEX voicemail_msgs_idx9 on voicemail_msgs(flags);

Just select uuid from voicemail_msgs where created_epoch < xyz and then use
the vm_delete API in fs_cli to zap the message:

vm_delete user at domain <uuid>

FS will do the rest: update the db, send out any MWI indications, etc.

If you get a working cron script let us know and we'll help you post it to
the wiki for all the world to see. :)

-MC


On Tue, Aug 28, 2012 at 2:04 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I don't think it exists. You could either custom-code a C module that
> polls at various times..
>
> Or just set a cron that queries the DB for old VMs then deletes the entry
> and the file.
>
> -Avi
>
>
>
> On Wed, Aug 29, 2012 at 12:00 AM, Chris Ferreira <chris at gonumina.com>wrote:
>
>> How can I set FreeSWITCH up so that OLD voicemails get deleted
>> automatically?
>>
>> I havent found anything in the Wiki or other documentation.
>>
>>
>>
>> Your input is much appreciated.
>>
>>
>>
>>
>> Thanks,
>>
>> -Chris
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From anthony.minessale at gmail.com  Wed Aug 29 10:34:36 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 29 Aug 2012 01:34:36 -0500
Subject: [Freeswitch-users] incoming call hangup if remote SDP contains
	a=crypto
In-Reply-To: <1346157540.11174.82.camel@marces.madrid.commsmundi.com>
References: <1346157540.11174.82.camel@marces.madrid.commsmundi.com>
Message-ID: <CAKbxfG8fm0iUXiMO2iR+VH4oK2oY2U8SXb8dU6QL+GtnwuLbMw@mail.gmail.com>

the problem is that you are using webrtc sipml5 and we don't have support
for webrtc sdp's yet.
It will work once we are done adding support.


On Tue, Aug 28, 2012 at 7:39 AM, Antonio <asilva at wirelessmundi.com> wrote:

> **
> Hi all,
>
> I'm having a problem to make calls, when the destination answer, the calls
> is hangup with incompatible destination.
>
> Debugging, i could see that is because fs drops the call, because it
> doesn't support ZRTP.
>
> So, is it normal that freeswitch hangs up the call when in the remote
> party send in the sdp "a=crypto:1 ... " ?
>
> Should it ignore and try to established the call using the supported
> methods in the sdp?
> Like for video calls, if the remote send video codecs, if we do not have
> them we don't hangup the call... just make a simple audio call...
>
>
> For incoming calls there is no problem.
>
>
> REMOTE SDP:
>
> 2012-08-28 10:37:56.068381 [DEBUG] sofia.c:6122 Remote SDP:
> v=0
> o=- 163879879 1 IN IP4 127.0.0.1
> s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)
> t=0 0
> a=group:BUNDLE audio video
> m=audio 57293 RTP/SAVPF 103 104 0 8 106 105 13 126
> c=IN IP4 80.28.139.221
> a=rtcp:57293 IN IP4 80.28.139.221
> a=candidate:3460286729 1 udp 2130714367 192.168.3.1 56196 typ host
> generation 0
> a=candidate:3460286729 2 udp 2130714367 192.168.3.1 56196 typ host
> generation 0
> a=candidate:1004190935 1 udp 2130714367 192.168.10.25 55956 typ host
> generation 0
> a=candidate:1004190935 2 udp 2130714367 192.168.10.25 55956 typ host
> generation 0
> a=candidate:1734362844 1 udp 1912610559 80.28.139.221 57293 typ srflx
> generation 0
> a=candidate:1734362844 2 udp 1912610559 80.28.139.221 57293 typ srflx
> generation 0
> a=candidate:3130178147 1 udp 1912610559 88.9.167.79 59009 typ srflx
> generation 0
> a=candidate:3130178147 2 udp 1912610559 88.9.167.79 59009 typ srflx
> generation 0
> a=candidate:2159818233 1 tcp 1694506751 192.168.3.1 52952 typ host
> generation 0
> a=candidate:2159818233 2 tcp 1694506751 192.168.3.1 52952 typ host
> generation 0
> a=candidate:1968783399 1 tcp 1694506751 192.168.10.25 45540 typ host
> generation 0
> a=candidate:1968783399 2 tcp 1694506751 192.168.10.25 45540 typ host
> generation 0
> a=ice-ufrag:SdZ956Hj4vXnNOhU
> a=ice-pwd:PQ9aJYFiLGiUfKZqqDXxQdtF
> a=mid:audio
> a=rtcp-mux
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:1j6B4cDFNA3Q99XoFujj3wYYex4ThJJNpOgbGifl
> a=rtpmap:103 ISAC/16000
> a=rtpmap:104 ISAC/32000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:106 CN/32000
> a=rtpmap:105 CN/16000
> a=rtpmap:13 CN/8000
> a=rtpmap:126 telephone-event/8000
> a=ssrc:2321878499 cname:qlN/YrHlo6j9iIJY
> a=ssrc:2321878499 mslabel:w4BkX3vCh3pGAdk7f2fqpUd6NT7JS2miWIhu
> a=ssrc:2321878499 label:w4BkX3vCh3pGAdk7f2fqpUd6NT7JS2miWIhu00
> m=video 57293 RTP/SAVPF 100 101 102
> c=IN IP4 80.28.139.221
> a=rtcp:57293 IN IP4 80.28.139.221
> a=candidate:3460286729 1 udp 2130714367 192.168.3.1 56196 typ host
> generation 0
> a=candidate:3460286729 2 udp 2130714367 192.168.3.1 56196 typ host
> generation 0
> a=candidate:1004190935 1 udp 2130714367 192.168.10.25 55956 typ host
> generation 0
> a=candidate:1004190935 2 udp 2130714367 192.168.10.25 55956 typ host
> generation 0
> a=candidate:1734362844 1 udp 1912610559 80.28.139.221 57293 typ srflx
> generation 0
> a=candidate:1734362844 2 udp 1912610559 80.28.139.221 57293 typ srflx
> generation 0
> a=candidate:3130178147 1 udp 1912610559 88.9.167.79 59009 typ srflx
> generation 0
> a=candidate:3130178147 2 udp 1912610559 88.9.167.79 59009 typ srflx
> generation 0
> a=candidate:2159818233 1 tcp 1694506751 192.168.3.1 52952 typ host
> generation 0
> a=candidate:2159818233 2 tcp 1694506751 192.168.3.1 52952 typ host
> generation 0
> a=candidate:1968783399 1 tcp 1694506751 192.168.10.25 45540 typ host
> generation 0
> a=candidate:1968783399 2 tcp 1694506751 192.168.10.25 45540 typ host
> generation 0
> a=ice-ufrag:SdZ956Hj4vXnNOhU
> a=ice-pwd:PQ9aJYFiLGiUfKZqqDXxQdtF
> a=mid:video
> a=rtcp-mux
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:1j6B4cDFNA3Q99XoFujj3wYYex4ThJJNpOgbGifl
> a=rtpmap:100 VP8/90000
> a=rtpmap:101 red/90000
> a=rtpmap:102 ulpfec/90000
>
>
>
>
> log when it fails:
>
> 2012-08-28 10:38:02.068442 [NOTICE] sofia.c:6853 Channel [sofia/
> 192.168.10.1/sip:2102 at 192.168.10.30:5062] has been answered
> 2012-08-28 10:38:02.068442 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash
> 2012-08-28 10:38:02.068442 [DEBUG] sofia_glue.c:3926 Deciding whether to
> pass zrtp-hash between legs
> 2012-08-28 10:38:02.068442 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ
> not set, so not propagating zrtp-hash
> 2012-08-28 10:38:02.068442 [ERR] sofia_glue.c:4837 a=crypto in RTP/AVP,
> refer to rfc3711
> 2012-08-28 10:38:02.068442 [DEBUG] switch_core_session.c:772 Send signal
> sofia/192.168.1.2_nat/2303 at test.commsmundi.com [BREAK]
> 2012-08-28 10:38:02.068442 [DEBUG] switch_channel.c:2924
> (sofia/192.168.1.2_nat/2303 at test.commsmundi.com) Callstate Change RINGING
> -> HANGUP
> 2012-08-28 10:38:02.068442 [NOTICE] switch_channel.c:3412 Hangup
> sofia/192.168.1.2_nat/2303 at test.commsmundi.com [CS_EXECUTE]
> [INCOMPATIBLE_DESTINATION]
>
>
>
>   --
> Best regards,
>
>  _________________________
>
> Ant?nio Silva
>
> E-mail:asilva at wirelessmundi.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From fdelawarde at wirelessmundi.com  Wed Aug 29 12:46:28 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 29 Aug 2012 10:46:28 +0200
Subject: [Freeswitch-users] Doubts about T38 passthru
Message-ID: <1346229988.23367.51.camel@luna.madrid.commsmundi.com>

Hi,

Whenever my provider puts me on hold, it reINVITES with an SDP
containing both audio (sendonly) and image media like so:

...
m=audio 5320 RTP/AVP 18 3 8 0 96
c=IN IP4 <ip_address>
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=sendonly
a=rtpmap:96 telephone-event/8000
m=image 5322 udptl t38
c=IN IP4 <ip_address>
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF


Not sure if it's a non-standard SDP, but that's what I get. Now when
t38_passthru=true, it only forwards the "image" part to the other leg.
When t38_passthru=false, it only forwards the "audio" part.

Is there a way to make t38_passthru forward the audio part as well? Or
should I just change to a better provider?

Thanks,
Fran?ois.




From fdelawarde at wirelessmundi.com  Wed Aug 29 12:48:32 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 29 Aug 2012 10:48:32 +0200
Subject: [Freeswitch-users] sip-router or opensips for good
 interoperability with freeswitch?
In-Reply-To: <0D1C698866F66045A6201FD0F59CAC90014678A962@EX.frontier.local>
References: <1345739304.23436.89.camel@luna.madrid.commsmundi.com>
	<50366E6E.4050902@newpace.ca>
	<0D1C698866F66045A6201FD0F59CAC90014678A962@EX.frontier.local>
Message-ID: <1346230112.23367.52.camel@luna.madrid.commsmundi.com>

Thanks for the comments, I'll just pick one at random as both seem to be
very good projects.

Fran?ois.

On Thu, 2012-08-23 at 14:46 -0400, Colin Mason wrote:
> I use OpenSIPS to route SIP to and from 30 or so FreeSWITCH virtual
> machines. It works flawlessly. I would recommend using 1.7.2 instead
> of the 1.8 branch as I have found it is not completely stable yet.
> 
>  
> 
> Colin
> 
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Adam Kelloway
> Sent: Thursday, August 23, 2012 1:55 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] sip-router or opensips for good
> interoperability with freeswitch?
> 
> 
>  
> 
> The are both pretty well functionally the same. You might find that
> Kamailo's .cfg file is easier to work with, as it allows you to name
> the routes (among other things), and it has a decent default cfg to
> use as a starting point. I have known people to have success with both
> opensips and kamailio.
> 
> On 23/08/2012 1:28 PM, Fran?ois Delawarde wrote:
> 
> 
>         Hi,
>          
>         Which of opensips or sip-router (kamailio) do you guys recommend to use
>         in front of freeswitch to create some big reliable cluster?
>          
>         I'm not sure if there is any huge difference as both seem to share a
>         same origin. Should I just throw a dice?, or is there one that is
>         clearly better than the other (or one I should avoid) in terms of
>         reliability / performance, or friendliness with FS?
>          
>         Thanks,
> 
>  
> 
> -- 
> Adam
> -- 
> 
> 
>  NewPace Logo
> 
> 
> 
> 
> 
> Adam Kelloway
> 
> 
> 
> Software
> Engineer,
> NewPace
> 
> 
>          phone
> 
> 
> +1 (902) 406?
> 8375  x1031
> 
> 
>          email
> 
> 
> Adam.Kelloway at NewPace.com
> 
> 
>        aim/msn
> 
> 
> Adam.Kelloway at NewPace.ca
> 
> 
> 
>  
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ntomer at newgen.co.in  Wed Aug 29 14:09:04 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Wed, 29 Aug 2012 15:39:04 +0530
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded to caller
	itself
Message-ID: <018601cd85ce$51c89730$f559c590$@co.in>

Hi,

 

I've configure mod_callcenter on Ubuntu 12.0.4. I've 5 extensions working on
6 soft phones - 1000 to 1005. 1000 is used to call the IVR number, rest 5
are agents.

 

The dial plan looks like -

 

<extension name="ivr_nitin">

      <condition field="destination_number" expression="^5002$">

        <action application="answer"/>

        <action application="sleep" data="500"/>

                <action application="ivr" data="nitin_ivr"/>

      </condition>

    </extension>

<!-- For Call Center -->

<!-- Newgen Support -->

       <extension>

          <condition field="destination_number" expression="^(450)$">

              <action application="set" data="caller_id_name=Newgen Support"
/>

              <action application="set" data="call_timeout=60" />

              <action application="set" data="originate_timeout=60" />

              <action application="callcenter" data="support at default"/>

           </condition>

        </extension>

 

<!-- Newgen Sales -->

        <extension>

           <condition field="destination_number" expression="^(451)$">

               <action application="set" data="caller_id_name=Newgen Sales"
/>

               <action application="set" data="call_timeout=60" />

               <action application="set" data="originate_timeout=60" />

               <action application="callcenter" data="sales at default"/>

           </condition>

        </extension>

 

<!-- Newgen Human Resources -->

        <extension>

           <condition field="destination_number" expression="^(452)$">

               <action application="set" data="caller_id_name=Newgen Human
Resources" />

                       <action application="set" data="call_timeout=60" />

               <action application="set" data="originate_timeout=60" />

               <action application="callcenter" data="hr at default"/>

           </condition>

        </extension>

 

IVR.xml -

 

<include>

        <menu name="nitin_ivr"

                greet-long="say:Welcome to Newgen Software Technologies
Limited. Press 1 for support, 2 for sales or 3 for human resource"

                greet-short="say:Welcome to Newgen. Press 1 for support, 2
for sales or 3 for human resources"

                invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"

                exit-sound="voicemail/vm-goodbye.wav"

                confirm-macro=""

                confirm-key=""

                tts-engine="flite"

                tts-voice="slt"

                confirm-attempts="3"

                timeout="3000"

                inter-digit-timeout="2000"

                max-failures="3"

                max-timeouts="3"

                digit-len="4">

 

                <entry action="menu-exec-app" digits="1" param="transfer 450
XML default"/>    <!-- Support -->

                <entry action="menu-exec-app" digits="2" param="transfer 451
XML default"/>    <!-- Sales -->

                <entry action="menu-exec-app" digits="3" param="transfer 452
XML default"/>    <!-- Human Resources -->

 

        </menu>

</include>

 

And callcenter.conf.xml -

 

<configuration name="callcenter.conf" description="CallCenter">

  <settings>

    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->

    <!--<param name="dbname" value="/dev/shm/callcenter.db"/>-->

  </settings>

 

    <queues>

                <queue name="sales at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/sales/${strftime(%Y-%m-%d-%H-%M-%S)}.${destin
ation_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="14400"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

                <queue name="support at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${dest
ination_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="60"/>

                                <param name="abandoned-resume-allowed"
value="false"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

                <queue name="hr at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${dest
ination_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="60"/>

                                <param name="abandoned-resume-allowed"
value="false"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

  </queues>

 

<!-- WARNING: Configuration of XML Agents will be updated into the DB upon
restart. -->

<!-- WARNING: Configuration of XML Tiers will reset the level and position
if those were supplied. -->

<!-- WARNING: Agents and Tiers XML config shouldn't be used in a multi FS
shared DB setup (Not currently supported anyway) -->

  <agents>

    <agent name="1001" type="callback" contact="[call_timeout=10]user/1001"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1002" type="callback" contact="[call_timeout=10]user/1002"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1003" type="callback" contact="[call_timeout=10]user/1003"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1004" type="callback" contact="[call_timeout=10]user/1004"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1005" type="callback" contact="[call_timeout=10]user/1005"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

  </agents>

  <tiers>

    <!-- If no level or position is provided, they will default to 1.  You
should do this to keep db value on restart. -->

                <!-- agent 1001 is in all three queues -->

                <tier agent="1001" queue="sales at default" level="1"
position="1"/>

                <tier agent="1001" queue="support at default" level="1"
position="1"/>

                <tier agent="1001" queue="hr at default" level="1"
position="1"/>

 

                <tier agent="1002" queue="sales at default" level="1"
position="1"/>

 

                <tier agent="1003" queue="support at default" level="1"
position="1"/>

 

                <tier agent="1004" queue="sales at default" level="1"
position="1"/>

                <tier agent="1004" queue="support at default" level="1"
position="1"/>

 

                <tier agent="1005" queue="hr at default" level="1"
position="1"/>

  </tiers>

 

</configuration>

 

When I call from extension 1000 and press 2 for sales, call is routed back
to extension 1000, which is not even in agents' list. Why is it happening?

 

Regards

 

Nitin

 


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From yufei.tao at redembedded.com  Wed Aug 29 14:18:09 2012
From: yufei.tao at redembedded.com (Yufei Tao)
Date: Wed, 29 Aug 2012 11:18:09 +0100
Subject: [Freeswitch-users] sip-router or opensips for good
In-Reply-To: <mailman.17877.1346234570.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.17877.1346234570.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <503DEC61.10607@redembedded.com>

If you use TCP/TLS a lot, Kamailio will save you a lot of grief as it's
got a much better TCP stack. I also find the cfg file is much easier to
work with as already mentioned.

Yufei

On 29/08/12 11:02, freeswitch-users-request at lists.freeswitch.org wrote:
> From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Adam Kelloway
> > Sent: Thursday, August 23, 2012 1:55 PM
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] sip-router or opensips for good
> > interoperability with freeswitch?
> >
> >
> >
> >
> > The are both pretty well functionally the same. You might find that
> > Kamailo's .cfg file is easier to work with, as it allows you to name
> > the routes (among other things), and it has a decent default cfg to
> > use as a starting point. I have known people to have success with both
> > opensips and kamailio.

--
Yufei Tao
Red Embedded

This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message.

You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender.

Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ


From daniel at pocock.com.au  Wed Aug 29 14:57:21 2012
From: daniel at pocock.com.au (Daniel Pocock)
Date: Wed, 29 Aug 2012 12:57:21 +0200
Subject: [Freeswitch-users] sip-router or opensips for good
In-Reply-To: <503DEC61.10607@redembedded.com>
References: <mailman.17877.1346234570.13443.freeswitch-users@lists.freeswitch.org>
	<503DEC61.10607@redembedded.com>
Message-ID: <503DF591.6040803@pocock.com.au>



I would suggest it is really a choice between Kamailio and repro (from
reSIProcate), which hasn't been mentioned in this thread, but is
actually very competitive and comes from a completely different
background (not rooted in SER)

Kamailio and repro are the two leaders when it comes to TLS and TCP
support, TLS is pretty much essential these days:
http://www.opentelecoms.org/federated-voip-tls

There are some very good tutorials about how to get started quickly with
either product:
http://www.opentelecoms.org/federated-voip-quick-start-howto



On 29/08/12 12:18, Yufei Tao wrote:
> If you use TCP/TLS a lot, Kamailio will save you a lot of grief as it's
> got a much better TCP stack. I also find the cfg file is much easier to
> work with as already mentioned.
> 
> Yufei
> 
> On 29/08/12 11:02, freeswitch-users-request at lists.freeswitch.org wrote:
>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>> Adam Kelloway
>>> Sent: Thursday, August 23, 2012 1:55 PM
>>> To: FreeSWITCH Users Help
>>> Subject: Re: [Freeswitch-users] sip-router or opensips for good
>>> interoperability with freeswitch?
>>>
>>>
>>>
>>>
>>> The are both pretty well functionally the same. You might find that
>>> Kamailo's .cfg file is easier to work with, as it allows you to name
>>> the routes (among other things), and it has a decent default cfg to
>>> use as a starting point. I have known people to have success with both
>>> opensips and kamailio.
> 
> --
> Yufei Tao
> Red Embedded
> 
> This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message.
> 
> You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender.
> 
> Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 daveh at beachdognet.com  Wed Aug 29 15:24:00 2012
From: daveh at beachdognet.com (Dave Horton)
Date: Wed, 29 Aug 2012 07:24:00 -0400
Subject: [Freeswitch-users] how to play a prompt after connection
Message-ID: <F62AB8C5-4184-4937-9AE0-27EF32F000ED@beachdognet.com>

> Set api_on_answer to execute
> uuid_broadcast<
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
> >
> which
> will let you pick which legs to play the recording to.
> You can pass in the UUID with ${uuid}
> 
> -Avi
> 
I tried this, and it worked.....sort of.  If I play the prompt the A leg it is fine -- caller hears the prompt, then is connected to called party.  If I play it to the B leg, however, the prompt does not seem to start playing at all until the B party actually starts talking, and then the prompt is played extremely choppy and slow.....it takes about 10 seconds to play a 4 second prompt.  Meanwhile, the caller and called party can't hear each other.  Any idea what is going on and how to fix it?  I am doing it in lua as follows:

session:execute("export","nolocal:jitterbuffer_msec=100") ;
session:setVariable("RECORD_STEREO","true") ;
session:setVariable("aleg_uuid",uuid) ;
session:execute("record_session", outfile) ;
session:execute("set","progress_timeout=60") ;
session:execute("set","call_timeout=120") ;
session:setVariable("hangup_after_bridge","true") ;
session:execute("bridge","sofia/normal_customer/${outdial}@${egress_gateway}") ;
--session:execute("bridge","{api_on_answer='uuid_broadcast " .. uuid .. " " .. this_call_is_being_recorded .. " " .. "bleg'}sofia/normal_customer/${outdial}@${egress_gateway}") ;



From miha at softnet.si  Wed Aug 29 17:27:49 2012
From: miha at softnet.si (Miha)
Date: Wed, 29 Aug 2012 15:27:49 +0200
Subject: [Freeswitch-users] group call pickup
Message-ID: <503E18D5.2020907@softnet.si>

Hi,

I have implemented call pickup, which is not working as it should.

When call comes in, user from same group can pick it up, but problem 
appears in this scenario:

A calls B. C do call pickup with *5 (in dialplan). OK this works (A is 
talking with C). But when D is calling E, if someone from same group 
pick this call, it do not pick D, it picks call A. Why? What I am doing 
wrong?

I hope you understand what I mean:D

my dialplan:

  <action application="set" 
data="called_party_callgroup=${user_data(${destination_number}.enterprise at enterprise.fs2.softnet.si 
var callgroup)}"/>
                 <action application="hash" 
data="insert/${destination_number}/${called_party_callgroup}/${uuid}"/>
                 <action application="hash" 
data="insert/last_dial/${called_party_callgroup}/${uuid}"/>


intercept:


        <extension name="group-intercept">
       <condition field="destination_number" expression="^\*5$">
         <action application="answer"/>
          <action application="intercept" 
data="${hash(select/last_dial/${callgroup})}"/>

         <action application="sleep" data="2000"/>
       </condition>
     </extension>

Thanks!
Miha


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From vipkilla at gmail.com  Wed Aug 29 17:49:58 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Wed, 29 Aug 2012 09:49:58 -0400
Subject: [Freeswitch-users] group call pickup
In-Reply-To: <503E18D5.2020907@softnet.si>
References: <503E18D5.2020907@softnet.si>
Message-ID: <CAC-LwPN4brMLWT5rxpUG--hiwUZz687SPq7C90yVf1BRZPQUJQ@mail.gmail.com>

That method of call pickup is out-dated, try using this, it works great:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pickup

On Wed, Aug 29, 2012 at 9:27 AM, Miha <miha at softnet.si> wrote:
> Hi,
>
> I have implemented call pickup, which is not working as it should.
>
> When call comes in, user from same group can pick it up, but problem appears
> in this scenario:
>
> A calls B. C do call pickup with *5 (in dialplan). OK this works (A is
> talking with C). But when D is calling E, if someone from same group pick
> this call, it do not pick D, it picks call A. Why? What I am doing wrong?
>
> I hope you understand what I mean:D
>
> my dialplan:
>
>  <action application="set"
> data="called_party_callgroup=${user_data(${destination_number}.enterprise at enterprise.fs2.softnet.si
> var callgroup)}"/>
>                 <action application="hash"
> data="insert/${destination_number}/${called_party_callgroup}/${uuid}"/>
>                 <action application="hash"
> data="insert/last_dial/${called_party_callgroup}/${uuid}"/>
>
>
> intercept:
>
>
>        <extension name="group-intercept">
>       <condition field="destination_number" expression="^\*5$">
>         <action application="answer"/>
>          <action application="intercept"
> data="${hash(select/last_dial/${callgroup})}"/>
>
>         <action application="sleep" data="2000"/>
>       </condition>
>     </extension>
>
> Thanks!
> Miha
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 alex at jajah.com  Wed Aug 29 17:57:23 2012
From: alex at jajah.com (Alex Massover)
Date: Wed, 29 Aug 2012 16:57:23 +0300
Subject: [Freeswitch-users] Prevent A leg from hangup after bridge with
	inbound ESL socket
Message-ID: <569384504C492C4580E88B5D54DFEAEA30CAFD33B0@jjex01.jajah.dublin>

Hi,

I have a very simple dialplan that just do park for incoming calls. All rest of leg management is done via ESL inbound socket.

I'm trying to do the same behavior like in this dialplan example, but from ESL inbound socket:
<action application="set" data="hangup_after_bridge=false"/>
   <action application="bridge" data="sofia/internal/1001@$${domain}"/>
   <action application="bridge" data="sofia/internal/1002@$${domain}"/>

The problem is with bridge API, if B leg doesn't answer (e.g. 404, or busy), A leg disconnects. But I'm trying to prevent A leg from disconnecting in order to do bridge to other place.

Looks like hangup_after_bridge=false, park_after_bridge=true, transfer_after_bridge etc don't have any effect when bridge done from inbound socket. A leg disconnects always.

Is there any way to keep A leg after bridge with inbound socket? I'm aware of originate, but prefer to user bridge.




--
Best Regards,
Alex Massover

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From miha at softnet.si  Wed Aug 29 18:45:05 2012
From: miha at softnet.si (Miha)
Date: Wed, 29 Aug 2012 16:45:05 +0200
Subject: [Freeswitch-users] group call pickup
In-Reply-To: <CAC-LwPN4brMLWT5rxpUG--hiwUZz687SPq7C90yVf1BRZPQUJQ@mail.gmail.com>
References: <503E18D5.2020907@softnet.si>
	<CAC-LwPN4brMLWT5rxpUG--hiwUZz687SPq7C90yVf1BRZPQUJQ@mail.gmail.com>
Message-ID: <503E2AF1.3030607@softnet.si>

Hi @Vik,

I am having problem with understanding how this works. After I bridge 
data I put like this:

  <anti-action application="bridge" 
data="[leg_timeout=30]user/${sip_to_user}.enterprise at xxx.xxx.xxx.xxx|user/${call_forwarding_number__nore}.enterprise at xxx.xxx.xxx.xxx, 
pickup/${callgroup
}"/>

To do pickup I am doing:


<extension name="group-intercept">
       <condition field="destination_number" expression="^\*5$">
<action application="pickup" data="${groupcall}"/>
          </condition>
     </extension>


What I am doing wrong or what I am missing?

Thanks!

Miha
On 8/29/2012 3:49 PM, Vik Killa wrote:
> That method of call pickup is out-dated, try using this, it works great:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pickup
>
> On Wed, Aug 29, 2012 at 9:27 AM, Miha <miha at softnet.si> wrote:
>> Hi,
>>
>> I have implemented call pickup, which is not working as it should.
>>
>> When call comes in, user from same group can pick it up, but problem appears
>> in this scenario:
>>
>> A calls B. C do call pickup with *5 (in dialplan). OK this works (A is
>> talking with C). But when D is calling E, if someone from same group pick
>> this call, it do not pick D, it picks call A. Why? What I am doing wrong?
>>
>> I hope you understand what I mean:D
>>
>> my dialplan:
>>
>>   <action application="set"
>> data="called_party_callgroup=${user_data(${destination_number}.enterprise at enterprise.fs2.softnet.si
>> var callgroup)}"/>
>>                  <action application="hash"
>> data="insert/${destination_number}/${called_party_callgroup}/${uuid}"/>
>>                  <action application="hash"
>> data="insert/last_dial/${called_party_callgroup}/${uuid}"/>
>>
>>
>> intercept:
>>
>>
>>         <extension name="group-intercept">
>>        <condition field="destination_number" expression="^\*5$">
>>          <action application="answer"/>
>>           <action application="intercept"
>> data="${hash(select/last_dial/${callgroup})}"/>
>>
>>          <action application="sleep" data="2000"/>
>>        </condition>
>>      </extension>
>>
>> Thanks!
>> Miha
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 roberto at i360tecnologia.com.br  Wed Aug 29 16:41:30 2012
From: roberto at i360tecnologia.com.br (Roberto Linck)
Date: Wed, 29 Aug 2012 09:41:30 -0300
Subject: [Freeswitch-users] Freeswitch - Stable Version
Message-ID: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>

Hello everybody,

I would like to know wich version of Freeswitch is the most stable one, in
your opinion and wich linux distro is the best choice to use with this
version.
I make this question because I need put a Freeswitch in production to
handle about 1500 concurrent calls. I know have the appropriate hardware,
but I really don't know the most indicated version and linux distro. I'm
using CentOS 6.2 but the newest version  of Freeswitch(1.2) doesn't work ok
with CentOS 6.x, so I'm opened to any option, since this can be stable.

I'll appreciate any contribution.



Roberto Linck
roberto at i360tecnologia.com.br
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From krice at freeswitch.org  Wed Aug 29 19:01:47 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 29 Aug 2012 10:01:47 -0500
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
References: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
Message-ID: <CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>

I used to use Centos6 and due to performance issues went to Debian 6 and
havent looked back...

at some point we really need to get some different people to oprofile their
freeswitch under their specific configs and give us the report logs so we
can see why centos6 is behaving so badly...

On Wed, Aug 29, 2012 at 7:41 AM, Roberto Linck <
roberto at i360tecnologia.com.br> wrote:

> Hello everybody,
>
> I would like to know wich version of Freeswitch is the most stable one, in
> your opinion and wich linux distro is the best choice to use with this
> version.
> I make this question because I need put a Freeswitch in production to
> handle about 1500 concurrent calls. I know have the appropriate hardware,
> but I really don't know the most indicated version and linux distro. I'm
> using CentOS 6.2 but the newest version  of Freeswitch(1.2) doesn't work ok
> with CentOS 6.x, so I'm opened to any option, since this can be stable.
>
> I'll appreciate any contribution.
>
>
>
> Roberto Linck
> roberto at i360tecnologia.com.br
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 vipkilla at gmail.com  Wed Aug 29 19:02:15 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Wed, 29 Aug 2012 11:02:15 -0400
Subject: [Freeswitch-users] group call pickup
In-Reply-To: <503E2AF1.3030607@softnet.si>
References: <503E18D5.2020907@softnet.si>
	<CAC-LwPN4brMLWT5rxpUG--hiwUZz687SPq7C90yVf1BRZPQUJQ@mail.gmail.com>
	<503E2AF1.3030607@softnet.si>
Message-ID: <CAC-LwPNHdfPwjpP5pkOOmvC7qWLuY4L288ca+LsqK9hZ5VvCyg@mail.gmail.com>

pickup/XXXX only needs to match here
<action application="pickup" data="XXXX"/>

You can use whatever variable as long as those two match.

On Wed, Aug 29, 2012 at 10:45 AM, Miha <miha at softnet.si> wrote:
> Hi @Vik,
>
> I am having problem with understanding how this works. After I bridge
> data I put like this:
>
>   <anti-action application="bridge"
> data="[leg_timeout=30]user/${sip_to_user}.enterprise at xxx.xxx.xxx.xxx|user/${call_forwarding_number__nore}.enterprise at xxx.xxx.xxx.xxx,
> pickup/${callgroup
> }"/>
>
> To do pickup I am doing:
>
>
> <extension name="group-intercept">
>        <condition field="destination_number" expression="^\*5$">
> <action application="pickup" data="${groupcall}"/>
>           </condition>
>      </extension>
>
>
> What I am doing wrong or what I am missing?
>
> Thanks!
>
> Miha
> On 8/29/2012 3:49 PM, Vik Killa wrote:
>> That method of call pickup is out-dated, try using this, it works great:
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pickup
>>
>> On Wed, Aug 29, 2012 at 9:27 AM, Miha <miha at softnet.si> wrote:
>>> Hi,
>>>
>>> I have implemented call pickup, which is not working as it should.
>>>
>>> When call comes in, user from same group can pick it up, but problem appears
>>> in this scenario:
>>>
>>> A calls B. C do call pickup with *5 (in dialplan). OK this works (A is
>>> talking with C). But when D is calling E, if someone from same group pick
>>> this call, it do not pick D, it picks call A. Why? What I am doing wrong?
>>>
>>> I hope you understand what I mean:D
>>>
>>> my dialplan:
>>>
>>>   <action application="set"
>>> data="called_party_callgroup=${user_data(${destination_number}.enterprise at enterprise.fs2.softnet.si
>>> var callgroup)}"/>
>>>                  <action application="hash"
>>> data="insert/${destination_number}/${called_party_callgroup}/${uuid}"/>
>>>                  <action application="hash"
>>> data="insert/last_dial/${called_party_callgroup}/${uuid}"/>
>>>
>>>
>>> intercept:
>>>
>>>
>>>         <extension name="group-intercept">
>>>        <condition field="destination_number" expression="^\*5$">
>>>          <action application="answer"/>
>>>           <action application="intercept"
>>> data="${hash(select/last_dial/${callgroup})}"/>
>>>
>>>          <action application="sleep" data="2000"/>
>>>        </condition>
>>>      </extension>
>>>
>>> Thanks!
>>> Miha
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 avi at avimarcus.net  Wed Aug 29 19:07:46 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 29 Aug 2012 18:07:46 +0300
Subject: [Freeswitch-users] endpoint /user and sofia_contact change?
Message-ID: <CALNZuwnv86enuhdOm57zSwF7G-D9U4gFnUBZR8R+QWjC5Wu4zw@mail.gmail.com>

I think I had 1.2 release yesterday and I would bridge to user/1000 or user
1000 at mydomain.com and/or do sofia_contact(1000) and all is fine.
However, since building on master in GIT last night: FreeSWITCH Version
1.3.0+git~20120828T230559Z~f1d201f03b (1.3.0; git at commit f1d201f03b on
Tue, 28 Aug 2012 23:05:59 Z)

They return not registered. I have to do sofia_contact internal/1000. I'm
not sure how to use user/ I replaced it with
${sofia_contact(internal/1000)}.
Was this an intentional change?
Is there some list of breaking changes somewhere?

-Avi
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From msc at freeswitch.org  Wed Aug 29 19:18:21 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Aug 2012 08:18:21 -0700
Subject: [Freeswitch-users] how to play a prompt after connection
In-Reply-To: <F62AB8C5-4184-4937-9AE0-27EF32F000ED@beachdognet.com>
References: <F62AB8C5-4184-4937-9AE0-27EF32F000ED@beachdognet.com>
Message-ID: <CAKzWOxUNeur5QdBdF9zHZriQdke86Pf+rPbZU_HCKzA6rNfe0Q@mail.gmail.com>

You could try doing a bridge_pre_execute and play the sounds to each leg
independently prior to the bridge. Check out these chan vars:
http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_aleg_app
http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_aleg_data
http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_bleg_app
http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_bleg_data

-MC

On Wed, Aug 29, 2012 at 4:24 AM, Dave Horton <daveh at beachdognet.com> wrote:

> > Set api_on_answer to execute
> > uuid_broadcast<
> > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
> > >
> > which
> > will let you pick which legs to play the recording to.
> > You can pass in the UUID with ${uuid}
> >
> > -Avi
> >
> I tried this, and it worked.....sort of.  If I play the prompt the A leg
> it is fine -- caller hears the prompt, then is connected to called party.
>  If I play it to the B leg, however, the prompt does not seem to start
> playing at all until the B party actually starts talking, and then the
> prompt is played extremely choppy and slow.....it takes about 10 seconds to
> play a 4 second prompt.  Meanwhile, the caller and called party can't hear
> each other.  Any idea what is going on and how to fix it?  I am doing it in
> lua as follows:
>
> session:execute("export","nolocal:jitterbuffer_msec=100") ;
> session:setVariable("RECORD_STEREO","true") ;
> session:setVariable("aleg_uuid",uuid) ;
> session:execute("record_session", outfile) ;
> session:execute("set","progress_timeout=60") ;
> session:execute("set","call_timeout=120") ;
> session:setVariable("hangup_after_bridge","true") ;
> session:execute("bridge","sofia/normal_customer/${outdial}@${egress_gateway}")
> ;
> --session:execute("bridge","{api_on_answer='uuid_broadcast " .. uuid .. "
> " .. this_call_is_being_recorded .. " " ..
> "bleg'}sofia/normal_customer/${outdial}@${egress_gateway}") ;
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Aug 29 19:20:31 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Aug 2012 08:20:31 -0700
Subject: [Freeswitch-users] Prevent A leg from hangup after bridge with
 inbound ESL socket
In-Reply-To: <569384504C492C4580E88B5D54DFEAEA30CAFD33B0@jjex01.jajah.dublin>
References: <569384504C492C4580E88B5D54DFEAEA30CAFD33B0@jjex01.jajah.dublin>
Message-ID: <CAKzWOxW+jCoFfd-e0GkBDvsez4f66xqDLnzpYM6-QesJ+3Md7g@mail.gmail.com>

Try this:
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail

-MC

On Wed, Aug 29, 2012 at 6:57 AM, Alex Massover <alex at jajah.com> wrote:

> Hi,****
>
> ** **
>
> I have a very simple dialplan that just do park for incoming calls. All
> rest of leg management is done via ESL inbound socket.****
>
> ** **
>
> I'm trying to do the same behavior like in this dialplan example, but from
> ESL inbound socket:****
>
> <action application="set" data="hangup_after_bridge=false"/> ****
>
>    <action application="bridge" data="sofia/internal/1001@$${domain}"/>***
> *
>
>    <action application="bridge" data="sofia/internal/1002@$${domain}"/>***
> *
>
> ** **
>
> The problem is with bridge API, if B leg doesn't answer (e.g. 404, or
> busy), A leg disconnects. But I'm trying to prevent A leg from
> disconnecting in order to do bridge to other place.****
>
> ** **
>
> Looks like hangup_after_bridge=false, park_after_bridge=true,
> transfer_after_bridge etc don't have any effect when bridge done from
> inbound socket. A leg disconnects always.****
>
> ** **
>
> Is there any way to keep A leg after bridge with inbound socket? I'm aware
> of originate, but prefer to user bridge.****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> --****
>
> Best Regards,****
>
> Alex Massover****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From miha at softnet.si  Wed Aug 29 19:22:42 2012
From: miha at softnet.si (Miha)
Date: Wed, 29 Aug 2012 17:22:42 +0200
Subject: [Freeswitch-users] group call pickup
In-Reply-To: <CAC-LwPNHdfPwjpP5pkOOmvC7qWLuY4L288ca+LsqK9hZ5VvCyg@mail.gmail.com>
References: <503E18D5.2020907@softnet.si>
	<CAC-LwPN4brMLWT5rxpUG--hiwUZz687SPq7C90yVf1BRZPQUJQ@mail.gmail.com>
	<503E2AF1.3030607@softnet.si>
	<CAC-LwPNHdfPwjpP5pkOOmvC7qWLuY4L288ca+LsqK9hZ5VvCyg@mail.gmail.com>
Message-ID: <503E33C2.2090007@softnet.si>

@Vik,

sorry for my stupid questions...

It is matched. As I am trying with account that are in the same callgroup.

log:

EXECUTE 
sofia/internal/018108754.enterprise at xxx.xxx.xxx.xxxbridge([leg_timeout=30]user/018108755.enterprise at enterprise.fs2.softnet.si|user/.enterprise at xxx.xxx.xxx.xxx, 
pickup/test)

afer *5:

Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Action 
pickup(${callgroup})

EXECUTE sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx pickup(test)

thanks!

Miha

On 8/29/2012 5:02 PM, Vik Killa wrote:
> pickup/XXXX only needs to match here
> <action application="pickup" data="XXXX"/>
>
> You can use whatever variable as long as those two match.
>
> On Wed, Aug 29, 2012 at 10:45 AM, Miha <miha at softnet.si> wrote:
>> Hi @Vik,
>>
>> I am having problem with understanding how this works. After I bridge
>> data I put like this:
>>
>>    <anti-action application="bridge"
>> data="[leg_timeout=30]user/${sip_to_user}.enterprise at xxx.xxx.xxx.xxx|user/${call_forwarding_number__nore}.enterprise at xxx.xxx.xxx.xxx,
>> pickup/${callgroup
>> }"/>
>>
>> To do pickup I am doing:
>>
>>
>> <extension name="group-intercept">
>>         <condition field="destination_number" expression="^\*5$">
>> <action application="pickup" data="${groupcall}"/>
>>            </condition>
>>       </extension>
>>
>>
>> What I am doing wrong or what I am missing?
>>
>> Thanks!
>>
>> Miha
>> On 8/29/2012 3:49 PM, Vik Killa wrote:
>>> That method of call pickup is out-dated, try using this, it works great:
>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pickup
>>>
>>> On Wed, Aug 29, 2012 at 9:27 AM, Miha <miha at softnet.si> wrote:
>>>> Hi,
>>>>
>>>> I have implemented call pickup, which is not working as it should.
>>>>
>>>> When call comes in, user from same group can pick it up, but problem appears
>>>> in this scenario:
>>>>
>>>> A calls B. C do call pickup with *5 (in dialplan). OK this works (A is
>>>> talking with C). But when D is calling E, if someone from same group pick
>>>> this call, it do not pick D, it picks call A. Why? What I am doing wrong?
>>>>
>>>> I hope you understand what I mean:D
>>>>
>>>> my dialplan:
>>>>
>>>>    <action application="set"
>>>> data="called_party_callgroup=${user_data(${destination_number}.enterprise at enterprise.fs2.softnet.si
>>>> var callgroup)}"/>
>>>>                   <action application="hash"
>>>> data="insert/${destination_number}/${called_party_callgroup}/${uuid}"/>
>>>>                   <action application="hash"
>>>> data="insert/last_dial/${called_party_callgroup}/${uuid}"/>
>>>>
>>>>
>>>> intercept:
>>>>
>>>>
>>>>          <extension name="group-intercept">
>>>>         <condition field="destination_number" expression="^\*5$">
>>>>           <action application="answer"/>
>>>>            <action application="intercept"
>>>> data="${hash(select/last_dial/${callgroup})}"/>
>>>>
>>>>           <action application="sleep" data="2000"/>
>>>>         </condition>
>>>>       </extension>
>>>>
>>>> Thanks!
>>>> Miha
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 vipkilla at gmail.com  Wed Aug 29 19:23:10 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Wed, 29 Aug 2012 11:23:10 -0400
Subject: [Freeswitch-users] endpoint /user and sofia_contact change?
In-Reply-To: <CALNZuwnv86enuhdOm57zSwF7G-D9U4gFnUBZR8R+QWjC5Wu4zw@mail.gmail.com>
References: <CALNZuwnv86enuhdOm57zSwF7G-D9U4gFnUBZR8R+QWjC5Wu4zw@mail.gmail.com>
Message-ID: <CAC-LwPMoGO2GCeMxC3Yw6-FV6wzwA6LvTyFKbmWt9O8s7Ua9pw@mail.gmail.com>

does 'sofia status profile <profile_name> reg' show them as registered?
try registering the handsets or flushing sofia registrations and
re-registering phone.

On Wed, Aug 29, 2012 at 11:07 AM, Avi Marcus <avi at avimarcus.net> wrote:
> I think I had 1.2 release yesterday and I would bridge to user/1000 or user
> 1000 at mydomain.com and/or do sofia_contact(1000) and all is fine.
> However, since building on master in GIT last night: FreeSWITCH Version
> 1.3.0+git~20120828T230559Z~f1d201f03b (1.3.0; git at commit f1d201f03b on
> Tue, 28 Aug 2012 23:05:59 Z)
>
> They return not registered. I have to do sofia_contact internal/1000. I'm
> not sure how to use user/ I replaced it with
> ${sofia_contact(internal/1000)}.
> Was this an intentional change?
> Is there some list of breaking changes somewhere?
>
> -Avi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 avi at avimarcus.net  Wed Aug 29 19:34:59 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 29 Aug 2012 18:34:59 +0300
Subject: [Freeswitch-users] endpoint /user and sofia_contact change?
In-Reply-To: <CAC-LwPMoGO2GCeMxC3Yw6-FV6wzwA6LvTyFKbmWt9O8s7Ua9pw@mail.gmail.com>
References: <CALNZuwnv86enuhdOm57zSwF7G-D9U4gFnUBZR8R+QWjC5Wu4zw@mail.gmail.com>
	<CAC-LwPMoGO2GCeMxC3Yw6-FV6wzwA6LvTyFKbmWt9O8s7Ua9pw@mail.gmail.com>
Message-ID: <CALNZuwmmm_3yo2tdYT_58M7Qgbz7QiRzmf42kpCwta5psD4YFg@mail.gmail.com>

freeswitch at default> sofia_contact 1000
error/user_not_registered

freeswitch at default> sofia_contact 1000 at sip.mydomain.com
error/user_not_registered

freeswitch at default> sofia_contact internal/1000 at sip.mydomain.com
sofia/internal/sip:1000 at 192.168.255.103:5077,sofia/internal/
sip:1000 at 192.168.255.103:5078,sofia/internal/sip:1000 at 192.168.255.103:32998
;ob

The first 2 worked before. I suppose this is a bug.

-Avi


On Wed, Aug 29, 2012 at 6:23 PM, Vik Killa <vipkilla at gmail.com> wrote:

> does 'sofia status profile <profile_name> reg' show them as registered?
> try registering the handsets or flushing sofia registrations and
> re-registering phone.
>
> On Wed, Aug 29, 2012 at 11:07 AM, Avi Marcus <avi at avimarcus.net> wrote:
> > I think I had 1.2 release yesterday and I would bridge to user/1000 or
> user
> > 1000 at mydomain.com and/or do sofia_contact(1000) and all is fine.
> > However, since building on master in GIT last night: FreeSWITCH Version
> > 1.3.0+git~20120828T230559Z~f1d201f03b (1.3.0; git at commit f1d201f03b on
> > Tue, 28 Aug 2012 23:05:59 Z)
> >
> > They return not registered. I have to do sofia_contact internal/1000. I'm
> > not sure how to use user/ I replaced it with
> > ${sofia_contact(internal/1000)}.
> > Was this an intentional change?
> > Is there some list of breaking changes somewhere?
> >
> > -Avi
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 kamminthang.nengzalam at a-cti.com  Wed Aug 29 19:46:22 2012
From: kamminthang.nengzalam at a-cti.com (Kamminthang Nengzalam)
Date: Wed, 29 Aug 2012 21:16:22 +0530
Subject: [Freeswitch-users] webrtc2sip
Message-ID: <CAL8RuBPmdVoPiC9hdsE+TAHVwiwdw=efEreY-2QKcfhudn-MHA@mail.gmail.com>

Hi Guys,

Can you lemme kno how to implement WebRTC in freeswitch
a really need it.

Thanks in advance for helping mi out.

-Kam
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From vbvbrj at gmail.com  Wed Aug 29 19:48:22 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Wed, 29 Aug 2012 18:48:22 +0300
Subject: [Freeswitch-users] uuid_broadcast muxing.
Message-ID: <503E39C6.40306@gmail.com>

Hello. I use uuid_broadcast to play a file to some leg:
api:executeString("uuid_broadcast "..caller_uuid.." 
phrase::queue-position,"..pos.." aleg")
But this pauses the moh file which is played to member waiting in 
callcenter. After two broadcasts, there is an error about leaking stream 
and adjusting of moh play. Is there a command to do the same but mux 
this playback of position with moh playing?

uuid_displace require absolute paths and also does not accept phrases. 
Also no leg specifying is accepted.

Thank you.


From covici at ccs.covici.com  Wed Aug 29 20:10:01 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 29 Aug 2012 12:10:01 -0400
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
References: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
	<CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
Message-ID: <12792.1346256601@ccs.covici.com>

Ken, I am unable to build on Debian squeeze -- it is having trouble with
create mod_amr.so.  
/usr/lib/gcc/i486-linux-gnu/4.4.5/../../../../lib/crt1.o: In function
`_start':
(.text+0x18): undefined reference to `main'

Should I file a Jira or what?

Ken Rice <krice at freeswitch.org> wrote:

> I used to use Centos6 and due to performance issues went to Debian 6 and
> havent looked back...
> 
> at some point we really need to get some different people to oprofile their
> freeswitch under their specific configs and give us the report logs so we
> can see why centos6 is behaving so badly...
> 
> On Wed, Aug 29, 2012 at 7:41 AM, Roberto Linck <
> roberto at i360tecnologia.com.br> wrote:
> 
> > Hello everybody,
> >
> > I would like to know wich version of Freeswitch is the most stable one, in
> > your opinion and wich linux distro is the best choice to use with this
> > version.
> > I make this question because I need put a Freeswitch in production to
> > handle about 1500 concurrent calls. I know have the appropriate hardware,
> > but I really don't know the most indicated version and linux distro. I'm
> > using CentOS 6.2 but the newest version  of Freeswitch(1.2) doesn't work ok
> > with CentOS 6.x, so I'm opened to any option, since this can be stable.
> >
> > I'll appreciate any contribution.
> >
> >
> >
> > Roberto Linck
> > roberto at i360tecnologia.com.br
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
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 msc at freeswitch.org  Wed Aug 29 20:09:45 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Aug 2012 09:09:45 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxXyEZWCvDfW=MEKvXcFG1hKJQdb=Xw8WOrQ_+i3HmgWHg@mail.gmail.com>

Hello all,

We are having a nice discussion today:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22

Mitch Capper and others who have experience will be talking to us about
TLS, certs, CA's and things like that. Hopefully we'll all be better
prepared to enable TLS transport on our devices and help those who come to
the IRC and ML with TLS questions.

Talk to you soon!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From krice at freeswitch.org  Wed Aug 29 20:18:04 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 29 Aug 2012 11:18:04 -0500
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <12792.1346256601@ccs.covici.com>
References: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
	<CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
	<12792.1346256601@ccs.covici.com>
Message-ID: <CAPcNkddN_LZj8pnhWYEc+DPEBCT9NT3UyTRGV+3M_xiOjKJeBA@mail.gmail.com>

are you sure?

I test daily on squeeze (i've switched to it as my primary platform)

mod_amr should build fine as its part of the default build from source.

unless you have patches in tree I would try the following,

git clean -fdx && git reset --hard

this will clean out any generated files etc...

git status to make sure you are on the v1.2.stable branch

then bootstrap, configure, and make as normal...

if you are still having issues, open a jira with a complete build log
attached


On Wed, Aug 29, 2012 at 11:10 AM, <covici at ccs.covici.com> wrote:

> Ken, I am unable to build on Debian squeeze -- it is having trouble with
> create mod_amr.so.
> /usr/lib/gcc/i486-linux-gnu/4.4.5/../../../../lib/crt1.o: In function
> `_start':
> (.text+0x18): undefined reference to `main'
>
> Should I file a Jira or what?
>
> Ken Rice <krice at freeswitch.org> wrote:
>
> > I used to use Centos6 and due to performance issues went to Debian 6 and
> > havent looked back...
> >
> > at some point we really need to get some different people to oprofile
> their
> > freeswitch under their specific configs and give us the report logs so we
> > can see why centos6 is behaving so badly...
> >
> > On Wed, Aug 29, 2012 at 7:41 AM, Roberto Linck <
> > roberto at i360tecnologia.com.br> wrote:
> >
> > > Hello everybody,
> > >
> > > I would like to know wich version of Freeswitch is the most stable
> one, in
> > > your opinion and wich linux distro is the best choice to use with this
> > > version.
> > > I make this question because I need put a Freeswitch in production to
> > > handle about 1500 concurrent calls. I know have the appropriate
> hardware,
> > > but I really don't know the most indicated version and linux distro.
> I'm
> > > using CentOS 6.2 but the newest version  of Freeswitch(1.2) doesn't
> work ok
> > > with CentOS 6.x, so I'm opened to any option, since this can be stable.
> > >
> > > I'll appreciate any contribution.
> > >
> > >
> > >
> > > Roberto Linck
> > > roberto at i360tecnologia.com.br
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> > ----------------------------------------------------
> > Alternatives:
> >
> > ----------------------------------------------------
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
> --
> 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mitch.capper at gmail.com  Wed Aug 29 20:20:12 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 29 Aug 2012 09:20:12 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
In-Reply-To: <CAKzWOxXyEZWCvDfW=MEKvXcFG1hKJQdb=Xw8WOrQ_+i3HmgWHg@mail.gmail.com>
References: <CAKzWOxXyEZWCvDfW=MEKvXcFG1hKJQdb=Xw8WOrQ_+i3HmgWHg@mail.gmail.com>
Message-ID: <CALn233o=FkQO-3D9Q64zkm4-xMdTP6+gGfTovDwGaNzN8aeFmg@mail.gmail.com>

Of course today is the 29th:)
~mitch

On Wed, Aug 29, 2012 at 9:09 AM, Michael Collins <msc at freeswitch.org> wrote:
> Hello all,
>
> We are having a nice discussion today:
>
> http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22
>
> Mitch Capper and others who have experience will be talking to us about TLS,
> certs, CA's and things like that. Hopefully we'll all be better prepared to
> enable TLS transport on our devices and help those who come to the IRC and
> ML with TLS questions.
>
> Talk to you soon!
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 slava at tangramltd.com  Wed Aug 29 20:33:39 2012
From: slava at tangramltd.com (Viacheslav Dubrovskyi)
Date: Wed, 29 Aug 2012 19:33:39 +0300
Subject: [Freeswitch-users] group call pickup
In-Reply-To: <CAC-LwPN4brMLWT5rxpUG--hiwUZz687SPq7C90yVf1BRZPQUJQ@mail.gmail.com>
References: <503E18D5.2020907@softnet.si>
	<CAC-LwPN4brMLWT5rxpUG--hiwUZz687SPq7C90yVf1BRZPQUJQ@mail.gmail.com>
Message-ID: <503E4463.6000107@tangramltd.com>

29.08.2012 16:49, Vik Killa ?????:
> That method of call pickup is out-dated, try using this, it works great:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pickup
Hi.

If I add pickup/12@$${domain} to fifo configuration

<configuration name="fifo.conf" description="FIFO Configuration">
<settings>
<param name="delete-all-outbound-member-on-startup" value="false"/>
</settings>
<fifos>
<fifo name="support_fifo@$${domain}" importance="0">
<member timeout="900" simo="10" 
lag="5">{member_wait=nowait}group/12,pickup/12@$${domain}</member>
</fifo>
</fifos>
</configuration>


Get in log
2012-08-29 16:20:23.121818 [WARNING] switch_ivr_originate.c:2356 Only 
calling the first element in the list in this mode.

And call only the first user of the group

How can I call pickup in this case?

-- 
WBR,
Viacheslav Dubrovskyi


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From covici at ccs.covici.com  Wed Aug 29 20:38:56 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 29 Aug 2012 12:38:56 -0400
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <CAPcNkddN_LZj8pnhWYEc+DPEBCT9NT3UyTRGV+3M_xiOjKJeBA@mail.gmail.com>
References: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
	<CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
	<12792.1346256601@ccs.covici.com>
	<CAPcNkddN_LZj8pnhWYEc+DPEBCT9NT3UyTRGV+3M_xiOjKJeBA@mail.gmail.com>
Message-ID: <17159.1346258336@ccs.covici.com>

Thanks -- that seems to have gotten past that!  Tricky stuff that a git
pull and checkout followed by bootstrap and configure would not have fix
things up.

Ken Rice <krice at freeswitch.org> wrote:

> are you sure?
> 
> I test daily on squeeze (i've switched to it as my primary platform)
> 
> mod_amr should build fine as its part of the default build from source.
> 
> unless you have patches in tree I would try the following,
> 
> git clean -fdx && git reset --hard
> 
> this will clean out any generated files etc...
> 
> git status to make sure you are on the v1.2.stable branch
> 
> then bootstrap, configure, and make as normal...
> 
> if you are still having issues, open a jira with a complete build log
> attached
> 
> 
> On Wed, Aug 29, 2012 at 11:10 AM, <covici at ccs.covici.com> wrote:
> 
> > Ken, I am unable to build on Debian squeeze -- it is having trouble with
> > create mod_amr.so.
> > /usr/lib/gcc/i486-linux-gnu/4.4.5/../../../../lib/crt1.o: In function
> > `_start':
> > (.text+0x18): undefined reference to `main'
> >
> > Should I file a Jira or what?
> >
> > Ken Rice <krice at freeswitch.org> wrote:
> >
> > > I used to use Centos6 and due to performance issues went to Debian 6 and
> > > havent looked back...
> > >
> > > at some point we really need to get some different people to oprofile
> > their
> > > freeswitch under their specific configs and give us the report logs so we
> > > can see why centos6 is behaving so badly...
> > >
> > > On Wed, Aug 29, 2012 at 7:41 AM, Roberto Linck <
> > > roberto at i360tecnologia.com.br> wrote:
> > >
> > > > Hello everybody,
> > > >
> > > > I would like to know wich version of Freeswitch is the most stable
> > one, in
> > > > your opinion and wich linux distro is the best choice to use with this
> > > > version.
> > > > I make this question because I need put a Freeswitch in production to
> > > > handle about 1500 concurrent calls. I know have the appropriate
> > hardware,
> > > > but I really don't know the most indicated version and linux distro.
> > I'm
> > > > using CentOS 6.2 but the newest version  of Freeswitch(1.2) doesn't
> > work ok
> > > > with CentOS 6.x, so I'm opened to any option, since this can be stable.
> > > >
> > > > I'll appreciate any contribution.
> > > >
> > > >
> > > >
> > > > Roberto Linck
> > > > roberto at i360tecnologia.com.br
> > > >
> > > >
> > _________________________________________________________________________
> > > > Professional FreeSWITCH Consulting Services:
> > > > consulting at freeswitch.org
> > > > http://www.freeswitchsolutions.com
> > > >
> > > > 
> > > > 
> > > >
> > > > Official FreeSWITCH Sites
> > > > http://www.freeswitch.org
> > > > http://wiki.freeswitch.org
> > > > http://www.cluecon.com
> > > >
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > > >
> > > >
> > >
> > > ----------------------------------------------------
> > > Alternatives:
> > >
> > > ----------------------------------------------------
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.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
> >
> > --
> > 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
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
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 krice at freeswitch.org  Wed Aug 29 20:47:09 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 29 Aug 2012 11:47:09 -0500
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <17159.1346258336@ccs.covici.com>
References: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
	<CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
	<12792.1346256601@ccs.covici.com>
	<CAPcNkddN_LZj8pnhWYEc+DPEBCT9NT3UyTRGV+3M_xiOjKJeBA@mail.gmail.com>
	<17159.1346258336@ccs.covici.com>
Message-ID: <CAPcNkde9C9Eety1LD5SE-b91A=1Btudoum2eMjPen9rsrhh4aA@mail.gmail.com>

who knows what happened there...

I have found when theres issues and using git, git clean and git reset
usually seem to fix things up

On Wed, Aug 29, 2012 at 11:38 AM, <covici at ccs.covici.com> wrote:

> Thanks -- that seems to have gotten past that!  Tricky stuff that a git
> pull and checkout followed by bootstrap and configure would not have fix
> things up.
>
> Ken Rice <krice at freeswitch.org> wrote:
>
> > are you sure?
> >
> > I test daily on squeeze (i've switched to it as my primary platform)
> >
> > mod_amr should build fine as its part of the default build from source.
> >
> > unless you have patches in tree I would try the following,
> >
> > git clean -fdx && git reset --hard
> >
> > this will clean out any generated files etc...
> >
> > git status to make sure you are on the v1.2.stable branch
> >
> > then bootstrap, configure, and make as normal...
> >
> > if you are still having issues, open a jira with a complete build log
> > attached
> >
> >
> > On Wed, Aug 29, 2012 at 11:10 AM, <covici at ccs.covici.com> wrote:
> >
> > > Ken, I am unable to build on Debian squeeze -- it is having trouble
> with
> > > create mod_amr.so.
> > > /usr/lib/gcc/i486-linux-gnu/4.4.5/../../../../lib/crt1.o: In function
> > > `_start':
> > > (.text+0x18): undefined reference to `main'
> > >
> > > Should I file a Jira or what?
> > >
> > > Ken Rice <krice at freeswitch.org> wrote:
> > >
> > > > I used to use Centos6 and due to performance issues went to Debian 6
> and
> > > > havent looked back...
> > > >
> > > > at some point we really need to get some different people to oprofile
> > > their
> > > > freeswitch under their specific configs and give us the report logs
> so we
> > > > can see why centos6 is behaving so badly...
> > > >
> > > > On Wed, Aug 29, 2012 at 7:41 AM, Roberto Linck <
> > > > roberto at i360tecnologia.com.br> wrote:
> > > >
> > > > > Hello everybody,
> > > > >
> > > > > I would like to know wich version of Freeswitch is the most stable
> > > one, in
> > > > > your opinion and wich linux distro is the best choice to use with
> this
> > > > > version.
> > > > > I make this question because I need put a Freeswitch in production
> to
> > > > > handle about 1500 concurrent calls. I know have the appropriate
> > > hardware,
> > > > > but I really don't know the most indicated version and linux
> distro.
> > > I'm
> > > > > using CentOS 6.2 but the newest version  of Freeswitch(1.2) doesn't
> > > work ok
> > > > > with CentOS 6.x, so I'm opened to any option, since this can be
> stable.
> > > > >
> > > > > I'll appreciate any contribution.
> > > > >
> > > > >
> > > > >
> > > > > Roberto Linck
> > > > > roberto at i360tecnologia.com.br
> > > > >
> > > > >
> > >
> _________________________________________________________________________
> > > > > Professional FreeSWITCH Consulting Services:
> > > > > consulting at freeswitch.org
> > > > > http://www.freeswitchsolutions.com
> > > > >
> > > > > 
> > > > > 
> > > > >
> > > > > Official FreeSWITCH Sites
> > > > > http://www.freeswitch.org
> > > > > http://wiki.freeswitch.org
> > > > > http://www.cluecon.com
> > > > >
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > UNSUBSCRIBE:
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > http://www.freeswitch.org
> > > > >
> > > > >
> > > >
> > > > ----------------------------------------------------
> > > > Alternatives:
> > > >
> > > > ----------------------------------------------------
> > > >
> _________________________________________________________________________
> > > > Professional FreeSWITCH Consulting Services:
> > > > consulting at freeswitch.org
> > > > http://www.freeswitchsolutions.com
> > > >
> > > > 
> > > > 
> > > >
> > > > Official FreeSWITCH Sites
> > > > http://www.freeswitch.org
> > > > http://wiki.freeswitch.org
> > > > http://www.cluecon.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
> > >
> > > --
> > > 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
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> > ----------------------------------------------------
> > Alternatives:
> >
> > ----------------------------------------------------
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
> --
> 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nasida at live.ru  Thu Aug 30 00:08:54 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Thu, 30 Aug 2012 00:08:54 +0400
Subject: [Freeswitch-users] [freeswitch] <defunct>
Message-ID: <SNT127-W65F6EBE2EAE8AA47841ABBD5A00@phx.gbl>


Hello guys,
What is the best way for understanding the reason why I begin to have [freeswitch] <defunct> process ?I don't use 'exec' directive.
CentOS release 6.3 (Final)FreeSWITCH Version 1.1.beta1 (git-de019ab 2012-05-03 15-23-57 +0200)
Please advise.Thanks.
 		 	   		  
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From nasida at live.ru  Thu Aug 30 00:38:03 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Thu, 30 Aug 2012 00:38:03 +0400
Subject: [Freeswitch-users] [freeswitch] <defunct>
In-Reply-To: <SNT127-W65F6EBE2EAE8AA47841ABBD5A00@phx.gbl>
References: <SNT127-W65F6EBE2EAE8AA47841ABBD5A00@phx.gbl>
Message-ID: <SNT127-W156C89CD0EFBFB0E5D9667D5A00@phx.gbl>


The answer here http://jira.freeswitch.org/browse/FS-3968 

Thanks.

From: nasida at live.ru
To: freeswitch-users at lists.freeswitch.org
Date: Thu, 30 Aug 2012 00:08:54 +0400
Subject: [Freeswitch-users] [freeswitch] <defunct>





Hello guys,
What is the best way for understanding the reason why I begin to have [freeswitch] <defunct> process ?I don't use 'exec' directive.
CentOS release 6.3 (Final)FreeSWITCH Version 1.1.beta1 (git-de019ab 2012-05-03 15-23-57 +0200)
Please advise.Thanks.
 		 	   		  

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 daniel.eiland at gmail.com  Thu Aug 30 00:18:05 2012
From: daniel.eiland at gmail.com (Daniel Eiland)
Date: Wed, 29 Aug 2012 16:18:05 -0400
Subject: [Freeswitch-users] RedHat 6.X Performance
Message-ID: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>

Hi folks,

We are currently running FreeSWITCH as a (sip/rtp) conference endpoint in
our system and recently completed some latency testing.  We found that
audio passing through FreeSWITCH was taking approx 40-60ms to be mixed and
sent out the other side.

I did some looking around and as far as I can tell, we don't have
FreeSWITCH configured with a jitter-buffer.  The only other possible
performance issue I could find was the 'tickless' issue (
http://wiki.freeswitch.org/wiki/Installation_Guide#Release.28es.29_6_and_Later)
when running FreeSWITCH on a RedHat 6+ installation.  However, its not
clear to me if the 'tickless' setting could cause this type of latency when
there are only two clients connected to FreeSWITCH.  I was wondering if
anybody else had experienced this type of problem or had any suggestions?

For reference, we are using a RTP/PCMU payload with 10ms samples.

Thanks,
Daniel
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From krice at freeswitch.org  Thu Aug 30 00:50:44 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 29 Aug 2012 15:50:44 -0500
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
Message-ID: <CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>

EL6/Centos 6 has been showing us some rather bad performance issues... if
you can run your config under oprofile and give us some of the outputs
related to freeswitch we would love to see this data and try to figure out
where the issue is and if its something we can work around or if its
something we need to look at getting theupstream to fix....

see http://oprofile.sourceforge.net/examples/ for how to use oprofile


On Wed, Aug 29, 2012 at 3:18 PM, Daniel Eiland <daniel.eiland at gmail.com>wrote:

> Hi folks,
>
> We are currently running FreeSWITCH as a (sip/rtp) conference endpoint in
> our system and recently completed some latency testing.  We found that
> audio passing through FreeSWITCH was taking approx 40-60ms to be mixed and
> sent out the other side.
>
> I did some looking around and as far as I can tell, we don't have
> FreeSWITCH configured with a jitter-buffer.  The only other possible
> performance issue I could find was the 'tickless' issue (
> http://wiki.freeswitch.org/wiki/Installation_Guide#Release.28es.29_6_and_Later)
> when running FreeSWITCH on a RedHat 6+ installation.  However, its not
> clear to me if the 'tickless' setting could cause this type of latency when
> there are only two clients connected to FreeSWITCH.  I was wondering if
> anybody else had experienced this type of problem or had any suggestions?
>
> For reference, we are using a RTP/PCMU payload with 10ms samples.
>
> Thanks,
> Daniel
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 vipkilla at gmail.com  Thu Aug 30 00:56:24 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Wed, 29 Aug 2012 16:56:24 -0400
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
Message-ID: <CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>

What IS the recommended linux distro to use for optimum performance?
I've heard debian, can someone be more specific though, like what
version, kernel, etc... ?

On Wed, Aug 29, 2012 at 4:50 PM, Ken Rice <krice at freeswitch.org> wrote:
> EL6/Centos 6 has been showing us some rather bad performance issues... if
> you can run your config under oprofile and give us some of the outputs
> related to freeswitch we would love to see this data and try to figure out
> where the issue is and if its something we can work around or if its
> something we need to look at getting theupstream to fix....
>
> see http://oprofile.sourceforge.net/examples/ for how to use oprofile
>
>
> On Wed, Aug 29, 2012 at 3:18 PM, Daniel Eiland <daniel.eiland at gmail.com>
> wrote:
>>
>> Hi folks,
>>
>> We are currently running FreeSWITCH as a (sip/rtp) conference endpoint in
>> our system and recently completed some latency testing.  We found that audio
>> passing through FreeSWITCH was taking approx 40-60ms to be mixed and sent
>> out the other side.
>>
>> I did some looking around and as far as I can tell, we don't have
>> FreeSWITCH configured with a jitter-buffer.  The only other possible
>> performance issue I could find was the 'tickless' issue
>> (http://wiki.freeswitch.org/wiki/Installation_Guide#Release.28es.29_6_and_Later)
>> when running FreeSWITCH on a RedHat 6+ installation.  However, its not clear
>> to me if the 'tickless' setting could cause this type of latency when there
>> are only two clients connected to FreeSWITCH.  I was wondering if anybody
>> else had experienced this type of problem or had any suggestions?
>>
>> For reference, we are using a RTP/PCMU payload with 10ms samples.
>>
>> Thanks,
>> Daniel
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Thu Aug 30 01:01:57 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Aug 2012 14:01:57 -0700
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
Message-ID: <CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>

On Wed, Aug 29, 2012 at 1:56 PM, Vik Killa <vipkilla at gmail.com> wrote:

> What IS the recommended linux distro to use for optimum performance?
> I've heard debian, can someone be more specific though, like what
> version, kernel, etc... ?
>

We try to stay out of the OS Holy Wars, so we don't have an "official"
recommended distro. However, I know more than a few people (disclosure: my
self included) who have been using Debian Squeeze 32 and 64 bit with zero
drama, zero issues, and solid performance.

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From krice at freeswitch.org  Thu Aug 30 01:09:05 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 29 Aug 2012 16:09:05 -0500
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
	<CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
Message-ID: <CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>

Centos5 worked well for me, Centos6 worked for crap... I went to debian 6
(squeeze) and problems went a way... using the latest stable on the debian6
line installing from the netinstall (non-free variant as I have hardware
thats not supported under the main netinstall iso) and then installing
freeswitch via the instructions on the wiki and no problems so far

(the centos6 issues were bad enough i moved my dev box to debian6)

On Wed, Aug 29, 2012 at 4:01 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 29, 2012 at 1:56 PM, Vik Killa <vipkilla at gmail.com> wrote:
>
>> What IS the recommended linux distro to use for optimum performance?
>> I've heard debian, can someone be more specific though, like what
>> version, kernel, etc... ?
>>
>
> We try to stay out of the OS Holy Wars, so we don't have an "official"
> recommended distro. However, I know more than a few people (disclosure: my
> self included) who have been using Debian Squeeze 32 and 64 bit with zero
> drama, zero issues, and solid performance.
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Thu Aug 30 02:03:16 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 29 Aug 2012 17:03:16 -0500
Subject: [Freeswitch-users] webrtc2sip
In-Reply-To: <CAL8RuBPmdVoPiC9hdsE+TAHVwiwdw=efEreY-2QKcfhudn-MHA@mail.gmail.com>
References: <CAL8RuBPmdVoPiC9hdsE+TAHVwiwdw=efEreY-2QKcfhudn-MHA@mail.gmail.com>
Message-ID: <CAKbxfG9tNGZSm=1P9X=4=t+xuGbc0YE2bx4+_NmFTZGVWYzx7Q@mail.gmail.com>

Work is underway for that.  If you want to contribute, use the paypal
button on the site.
Otherwise you will need to wait for it to be completed.


On Wed, Aug 29, 2012 at 10:46 AM, Kamminthang Nengzalam
<kamminthang.nengzalam at a-cti.com> wrote:
> Hi Guys,
>
> Can you lemme kno how to implement WebRTC in freeswitch
> a really need it.
>
> Thanks in advance for helping mi out.
>
> -Kam
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From asaad2 at gmail.com  Thu Aug 30 02:21:03 2012
From: asaad2 at gmail.com (BookBag)
Date: Wed, 29 Aug 2012 17:21:03 -0500
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
	<CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
	<CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
Message-ID: <CAAdTMa-keR12qGRZOQdOijR8PAZU3GyGORcT7ABNnfoJbARdfA@mail.gmail.com>

Woe is me. I'm using centos 6 and I figured that performance issues was due
to my cheap box and was actually planning to buy a more powerful computer.

What specific issues did you guys encounter and has anyone tried the new
6.3 version. I would like to keep an eye out for these problems on my
distro.
On Aug 29, 2012 5:10 PM, "Ken Rice" <krice at freeswitch.org> wrote:

> Centos5 worked well for me, Centos6 worked for crap... I went to debian 6
> (squeeze) and problems went a way... using the latest stable on the debian6
> line installing from the netinstall (non-free variant as I have hardware
> thats not supported under the main netinstall iso) and then installing
> freeswitch via the instructions on the wiki and no problems so far
>
> (the centos6 issues were bad enough i moved my dev box to debian6)
>
> On Wed, Aug 29, 2012 at 4:01 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Wed, Aug 29, 2012 at 1:56 PM, Vik Killa <vipkilla at gmail.com> wrote:
>>
>>> What IS the recommended linux distro to use for optimum performance?
>>> I've heard debian, can someone be more specific though, like what
>>> version, kernel, etc... ?
>>>
>>
>> We try to stay out of the OS Holy Wars, so we don't have an "official"
>> recommended distro. However, I know more than a few people (disclosure: my
>> self included) who have been using Debian Squeeze 32 and 64 bit with zero
>> drama, zero issues, and solid performance.
>>
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jaybinks at gmail.com  Thu Aug 30 02:26:42 2012
From: jaybinks at gmail.com (jay binks)
Date: Thu, 30 Aug 2012 08:26:42 +1000
Subject: [Freeswitch-users] webrtc2sip
In-Reply-To: <CAKbxfG9tNGZSm=1P9X=4=t+xuGbc0YE2bx4+_NmFTZGVWYzx7Q@mail.gmail.com>
References: <CAL8RuBPmdVoPiC9hdsE+TAHVwiwdw=efEreY-2QKcfhudn-MHA@mail.gmail.com>
	<CAKbxfG9tNGZSm=1P9X=4=t+xuGbc0YE2bx4+_NmFTZGVWYzx7Q@mail.gmail.com>
Message-ID: <CAO47Hpf5T7yKc=9RA=vNKd=_yqfrDCmecL41pkO+hk+DMDGvkQ@mail.gmail.com>

excuse my ignorance, but will webrtc2sip support video, or audio only at
this stage ?

Jay


On 30 August 2012 08:03, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> Work is underway for that.  If you want to contribute, use the paypal
> button on the site.
> Otherwise you will need to wait for it to be completed.
>
>
> On Wed, Aug 29, 2012 at 10:46 AM, Kamminthang Nengzalam
> <kamminthang.nengzalam at a-cti.com> wrote:
> > Hi Guys,
> >
> > Can you lemme kno how to implement WebRTC in freeswitch
> > a really need it.
> >
> > Thanks in advance for helping mi out.
> >
> > -Kam
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

Jay
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From anthony.minessale at gmail.com  Thu Aug 30 02:56:39 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 29 Aug 2012 17:56:39 -0500
Subject: [Freeswitch-users] webrtc2sip
In-Reply-To: <CAO47Hpf5T7yKc=9RA=vNKd=_yqfrDCmecL41pkO+hk+DMDGvkQ@mail.gmail.com>
References: <CAL8RuBPmdVoPiC9hdsE+TAHVwiwdw=efEreY-2QKcfhudn-MHA@mail.gmail.com>
	<CAKbxfG9tNGZSm=1P9X=4=t+xuGbc0YE2bx4+_NmFTZGVWYzx7Q@mail.gmail.com>
	<CAO47Hpf5T7yKc=9RA=vNKd=_yqfrDCmecL41pkO+hk+DMDGvkQ@mail.gmail.com>
Message-ID: <CAKbxfG9PjN6uP+2PDka0v+t_0Ce9LNLFHdZG3HX=e3+6A1SMrg@mail.gmail.com>

once its in place it should do both

On Wed, Aug 29, 2012 at 5:26 PM, jay binks <jaybinks at gmail.com> wrote:
> excuse my ignorance, but will webrtc2sip support video, or audio only at
> this stage ?
>
> Jay
>
>
> On 30 August 2012 08:03, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>>
>> Work is underway for that.  If you want to contribute, use the paypal
>> button on the site.
>> Otherwise you will need to wait for it to be completed.
>>
>>
>> On Wed, Aug 29, 2012 at 10:46 AM, Kamminthang Nengzalam
>> <kamminthang.nengzalam at a-cti.com> wrote:
>> > Hi Guys,
>> >
>> > Can you lemme kno how to implement WebRTC in freeswitch
>> > a really need it.
>> >
>> > Thanks in advance for helping mi out.
>> >
>> > -Kam
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Thu Aug 30 03:21:08 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 29 Aug 2012 18:21:08 -0500
Subject: [Freeswitch-users] uuid_broadcast muxing.
In-Reply-To: <503E39C6.40306@gmail.com>
References: <503E39C6.40306@gmail.com>
Message-ID: <CAKbxfG9CsU4U+xEYmVBWrGdXDDdsSfuXQEwQW7HP5bJQtogLGA@mail.gmail.com>

The error is harmless warning to remind you the local_stream is being ignored.


On Wed, Aug 29, 2012 at 10:48 AM, Vbvbrj <vbvbrj at gmail.com> wrote:
> Hello. I use uuid_broadcast to play a file to some leg:
> api:executeString("uuid_broadcast "..caller_uuid.."
> phrase::queue-position,"..pos.." aleg")
> But this pauses the moh file which is played to member waiting in
> callcenter. After two broadcasts, there is an error about leaking stream
> and adjusting of moh play. Is there a command to do the same but mux
> this playback of position with moh playing?
>
> uuid_displace require absolute paths and also does not accept phrases.
> Also no leg specifying is accepted.
>
> Thank you.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

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

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

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


From roberto at i360tecnologia.com.br  Thu Aug 30 03:56:25 2012
From: roberto at i360tecnologia.com.br (Roberto Linck)
Date: Wed, 29 Aug 2012 20:56:25 -0300
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
References: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
	<CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
Message-ID: <CAFTa0ycv_pTjCfXL7tYT5vcx=jad68_r+tcRdnC84QHXnr4nVA@mail.gmail.com>

Ken,

Your feedback about using Debian and "...havent looked back." made the
difference to me.
I'll try Debian and Freeswitch.
Thank you.

2012/8/29 Ken Rice <krice at freeswitch.org>

> I used to use Centos6 and due to performance issues went to Debian 6 and
> havent looked back...
>
> at some point we really need to get some different people to oprofile
> their freeswitch under their specific configs and give us the report logs
> so we can see why centos6 is behaving so badly...
>
> On Wed, Aug 29, 2012 at 7:41 AM, Roberto Linck <
> roberto at i360tecnologia.com.br> wrote:
>
>> Hello everybody,
>>
>> I would like to know wich version of Freeswitch is the most stable one,
>> in your opinion and wich linux distro is the best choice to use with this
>> version.
>> I make this question because I need put a Freeswitch in production to
>> handle about 1500 concurrent calls. I know have the appropriate hardware,
>> but I really don't know the most indicated version and linux distro. I'm
>> using CentOS 6.2 but the newest version  of Freeswitch(1.2) doesn't work ok
>> with CentOS 6.x, so I'm opened to any option, since this can be stable.
>>
>> I'll appreciate any contribution.
>>
>>
>>
>> Roberto Linck
>> roberto at i360tecnologia.com.br
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 kris at kriskinc.com  Thu Aug 30 04:30:30 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 29 Aug 2012 20:30:30 -0400
Subject: [Freeswitch-users] webrtc2sip
In-Reply-To: <CAKbxfG9PjN6uP+2PDka0v+t_0Ce9LNLFHdZG3HX=e3+6A1SMrg@mail.gmail.com>
References: <CAL8RuBPmdVoPiC9hdsE+TAHVwiwdw=efEreY-2QKcfhudn-MHA@mail.gmail.com>
	<CAKbxfG9tNGZSm=1P9X=4=t+xuGbc0YE2bx4+_NmFTZGVWYzx7Q@mail.gmail.com>
	<CAO47Hpf5T7yKc=9RA=vNKd=_yqfrDCmecL41pkO+hk+DMDGvkQ@mail.gmail.com>
	<CAKbxfG9PjN6uP+2PDka0v+t_0Ce9LNLFHdZG3HX=e3+6A1SMrg@mail.gmail.com>
Message-ID: <CAKDfjgcQA7JYABtCWAR6Q3xNd5fxAkKMW32uu4LMywV3m9dEsw@mail.gmail.com>

I hate to keep asking, but what about STUN/TURN/ICE?

On Wed, Aug 29, 2012 at 6:56 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> once its in place it should do both
>
> On Wed, Aug 29, 2012 at 5:26 PM, jay binks <jaybinks at gmail.com> wrote:
>> excuse my ignorance, but will webrtc2sip support video, or audio only at
>> this stage ?
>>
>> Jay
>>
>>
>> On 30 August 2012 08:03, Anthony Minessale <anthony.minessale at gmail.com>
>> wrote:
>>>
>>> Work is underway for that.  If you want to contribute, use the paypal
>>> button on the site.
>>> Otherwise you will need to wait for it to be completed.
>>>
>>>
>>> On Wed, Aug 29, 2012 at 10:46 AM, Kamminthang Nengzalam
>>> <kamminthang.nengzalam at a-cti.com> wrote:
>>> > Hi Guys,
>>> >
>>> > Can you lemme kno how to implement WebRTC in freeswitch
>>> > a really need it.
>>> >
>>> > Thanks in advance for helping mi out.
>>> >
>>> > -Kam
>>> >
>>> >
>>> > _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>
>> Jay
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Kristian Kielhofner


From anthony.minessale at gmail.com  Thu Aug 30 05:36:16 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 29 Aug 2012 20:36:16 -0500
Subject: [Freeswitch-users] webrtc2sip
In-Reply-To: <CAKDfjgcQA7JYABtCWAR6Q3xNd5fxAkKMW32uu4LMywV3m9dEsw@mail.gmail.com>
References: <CAL8RuBPmdVoPiC9hdsE+TAHVwiwdw=efEreY-2QKcfhudn-MHA@mail.gmail.com>
	<CAKbxfG9tNGZSm=1P9X=4=t+xuGbc0YE2bx4+_NmFTZGVWYzx7Q@mail.gmail.com>
	<CAO47Hpf5T7yKc=9RA=vNKd=_yqfrDCmecL41pkO+hk+DMDGvkQ@mail.gmail.com>
	<CAKbxfG9PjN6uP+2PDka0v+t_0Ce9LNLFHdZG3HX=e3+6A1SMrg@mail.gmail.com>
	<CAKDfjgcQA7JYABtCWAR6Q3xNd5fxAkKMW32uu4LMywV3m9dEsw@mail.gmail.com>
Message-ID: <CAKbxfG-nG24DMwA9tsDYsUmX2gOHoVR7zRAfP+O1ZUXPA73Qjw@mail.gmail.com>

the stun / ice stuff is mandatory so ya.  turn I have no exp with.
you need stun/ice/srtp as a prereq.  There are lot of changes to do
into to core to refactor and avoid code dup and do it right that come
before the new module its a giant undertaking so we are looking for
sponsors.


On Wed, Aug 29, 2012 at 7:30 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> I hate to keep asking, but what about STUN/TURN/ICE?
>
> On Wed, Aug 29, 2012 at 6:56 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> once its in place it should do both
>>
>> On Wed, Aug 29, 2012 at 5:26 PM, jay binks <jaybinks at gmail.com> wrote:
>>> excuse my ignorance, but will webrtc2sip support video, or audio only at
>>> this stage ?
>>>
>>> Jay
>>>
>>>
>>> On 30 August 2012 08:03, Anthony Minessale <anthony.minessale at gmail.com>
>>> wrote:
>>>>
>>>> Work is underway for that.  If you want to contribute, use the paypal
>>>> button on the site.
>>>> Otherwise you will need to wait for it to be completed.
>>>>
>>>>
>>>> On Wed, Aug 29, 2012 at 10:46 AM, Kamminthang Nengzalam
>>>> <kamminthang.nengzalam at a-cti.com> wrote:
>>>> > Hi Guys,
>>>> >
>>>> > Can you lemme kno how to implement WebRTC in freeswitch
>>>> > a really need it.
>>>> >
>>>> > Thanks in advance for helping mi out.
>>>> >
>>>> > -Kam
>>>> >
>>>> >
>>>> > _________________________________________________________________________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > 
>>>> > 
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://wiki.freeswitch.org
>>>> > http://www.cluecon.com
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>
>>> Jay
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Kristian Kielhofner
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

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

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

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


From bdfoster at endigotech.com  Thu Aug 30 05:59:52 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 29 Aug 2012 21:59:52 -0400
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <CAPcNkde9C9Eety1LD5SE-b91A=1Btudoum2eMjPen9rsrhh4aA@mail.gmail.com>
References: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
	<CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
	<12792.1346256601@ccs.covici.com>
	<CAPcNkddN_LZj8pnhWYEc+DPEBCT9NT3UyTRGV+3M_xiOjKJeBA@mail.gmail.com>
	<17159.1346258336@ccs.covici.com>
	<CAPcNkde9C9Eety1LD5SE-b91A=1Btudoum2eMjPen9rsrhh4aA@mail.gmail.com>
Message-ID: <CAKV-jFwetO7bc4OoRjovtmVFRDN0sq7UAQpi6czBYwvqxb3RoQ@mail.gmail.com>

Sorry I have to do this...

Go Debian!!!

Brian Foster
Endigo Computer LLC

Sent from a mobile device.
On Aug 29, 2012 12:49 PM, "Ken Rice" <krice at freeswitch.org> wrote:

> who knows what happened there...
>
> I have found when theres issues and using git, git clean and git reset
> usually seem to fix things up
>
> On Wed, Aug 29, 2012 at 11:38 AM, <covici at ccs.covici.com> wrote:
>
>> Thanks -- that seems to have gotten past that!  Tricky stuff that a git
>> pull and checkout followed by bootstrap and configure would not have fix
>> things up.
>>
>> Ken Rice <krice at freeswitch.org> wrote:
>>
>> > are you sure?
>> >
>> > I test daily on squeeze (i've switched to it as my primary platform)
>> >
>> > mod_amr should build fine as its part of the default build from source.
>> >
>> > unless you have patches in tree I would try the following,
>> >
>> > git clean -fdx && git reset --hard
>> >
>> > this will clean out any generated files etc...
>> >
>> > git status to make sure you are on the v1.2.stable branch
>> >
>> > then bootstrap, configure, and make as normal...
>> >
>> > if you are still having issues, open a jira with a complete build log
>> > attached
>> >
>> >
>> > On Wed, Aug 29, 2012 at 11:10 AM, <covici at ccs.covici.com> wrote:
>> >
>> > > Ken, I am unable to build on Debian squeeze -- it is having trouble
>> with
>> > > create mod_amr.so.
>> > > /usr/lib/gcc/i486-linux-gnu/4.4.5/../../../../lib/crt1.o: In function
>> > > `_start':
>> > > (.text+0x18): undefined reference to `main'
>> > >
>> > > Should I file a Jira or what?
>> > >
>> > > Ken Rice <krice at freeswitch.org> wrote:
>> > >
>> > > > I used to use Centos6 and due to performance issues went to Debian
>> 6 and
>> > > > havent looked back...
>> > > >
>> > > > at some point we really need to get some different people to
>> oprofile
>> > > their
>> > > > freeswitch under their specific configs and give us the report logs
>> so we
>> > > > can see why centos6 is behaving so badly...
>> > > >
>> > > > On Wed, Aug 29, 2012 at 7:41 AM, Roberto Linck <
>> > > > roberto at i360tecnologia.com.br> wrote:
>> > > >
>> > > > > Hello everybody,
>> > > > >
>> > > > > I would like to know wich version of Freeswitch is the most stable
>> > > one, in
>> > > > > your opinion and wich linux distro is the best choice to use with
>> this
>> > > > > version.
>> > > > > I make this question because I need put a Freeswitch in
>> production to
>> > > > > handle about 1500 concurrent calls. I know have the appropriate
>> > > hardware,
>> > > > > but I really don't know the most indicated version and linux
>> distro.
>> > > I'm
>> > > > > using CentOS 6.2 but the newest version  of Freeswitch(1.2)
>> doesn't
>> > > work ok
>> > > > > with CentOS 6.x, so I'm opened to any option, since this can be
>> stable.
>> > > > >
>> > > > > I'll appreciate any contribution.
>> > > > >
>> > > > >
>> > > > >
>> > > > > Roberto Linck
>> > > > > roberto at i360tecnologia.com.br
>> > > > >
>> > > > >
>> > >
>> _________________________________________________________________________
>> > > > > Professional FreeSWITCH Consulting Services:
>> > > > > consulting at freeswitch.org
>> > > > > http://www.freeswitchsolutions.com
>> > > > >
>> > > > > 
>> > > > > 
>> > > > >
>> > > > > Official FreeSWITCH Sites
>> > > > > http://www.freeswitch.org
>> > > > > http://wiki.freeswitch.org
>> > > > > http://www.cluecon.com
>> > > > >
>> > > > > FreeSWITCH-users mailing list
>> > > > > FreeSWITCH-users at lists.freeswitch.org
>> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > > > UNSUBSCRIBE:
>> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > > > http://www.freeswitch.org
>> > > > >
>> > > > >
>> > > >
>> > > > ----------------------------------------------------
>> > > > Alternatives:
>> > > >
>> > > > ----------------------------------------------------
>> > > >
>> _________________________________________________________________________
>> > > > Professional FreeSWITCH Consulting Services:
>> > > > consulting at freeswitch.org
>> > > > http://www.freeswitchsolutions.com
>> > > >
>> > > > 
>> > > > 
>> > > >
>> > > > Official FreeSWITCH Sites
>> > > > http://www.freeswitch.org
>> > > > http://wiki.freeswitch.org
>> > > > http://www.cluecon.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
>> > >
>> > > --
>> > > 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
>> > >
>> > >
>> _________________________________________________________________________
>> > > Professional FreeSWITCH Consulting Services:
>> > > consulting at freeswitch.org
>> > > http://www.freeswitchsolutions.com
>> > >
>> > > 
>> > > 
>> > >
>> > > Official FreeSWITCH Sites
>> > > http://www.freeswitch.org
>> > > http://wiki.freeswitch.org
>> > > http://www.cluecon.com
>> > >
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> > >
>> >
>> > ----------------------------------------------------
>> > Alternatives:
>> >
>> > ----------------------------------------------------
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.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
>>
>> --
>> 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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jaybinks at gmail.com  Thu Aug 30 06:12:06 2012
From: jaybinks at gmail.com (jay binks)
Date: Thu, 30 Aug 2012 12:12:06 +1000
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <CAKV-jFwetO7bc4OoRjovtmVFRDN0sq7UAQpi6czBYwvqxb3RoQ@mail.gmail.com>
References: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
	<CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
	<12792.1346256601@ccs.covici.com>
	<CAPcNkddN_LZj8pnhWYEc+DPEBCT9NT3UyTRGV+3M_xiOjKJeBA@mail.gmail.com>
	<17159.1346258336@ccs.covici.com>
	<CAPcNkde9C9Eety1LD5SE-b91A=1Btudoum2eMjPen9rsrhh4aA@mail.gmail.com>
	<CAKV-jFwetO7bc4OoRjovtmVFRDN0sq7UAQpi6czBYwvqxb3RoQ@mail.gmail.com>
Message-ID: <CAO47HpcbUzsQw=jJh8rK1bS0mM3Sm=qgwaR4zx7mRB-==JYKGA@mail.gmail.com>

Ive been using Debian for years and its performed great.

another thumbs up for freeswitch on Debian 6
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From krice at freeswitch.org  Thu Aug 30 07:17:36 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 29 Aug 2012 22:17:36 -0500
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <CAFTa0ycv_pTjCfXL7tYT5vcx=jad68_r+tcRdnC84QHXnr4nVA@mail.gmail.com>
References: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
	<CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
	<CAFTa0ycv_pTjCfXL7tYT5vcx=jad68_r+tcRdnC84QHXnr4nVA@mail.gmail.com>
Message-ID: <CAPcNkdefcnY+_By3y1nUS8H=9R48jq4m2fmnqaDc=5sNCK2xUw@mail.gmail.com>

Just keep in mind thats my personal experience on the subject...

Various people are still running Centos5.8, Arch, ${some_other_distro} ...

The best advise I can offer to anyone reading these follow ups, is pick the
tool that works properly for you to accomplish the task at hand. For some
FreeSWITCH users, this even means using a MicroSoft Windows platform.

And by all means test as close to production as you can before actually
deploying...

chances are, pretty much whatever platform you pick will work unless you
are doing something really crazy (like trying to cram an OC12 worth of
calls through 1 machine

(why you would want to risk that much business on one machine I am not
sure, but I am sure someone somewhere will try it eventually)

K


On Wed, Aug 29, 2012 at 6:56 PM, Roberto Linck <
roberto at i360tecnologia.com.br> wrote:

> Ken,
>
> Your feedback about using Debian and "...havent looked back." made the
> difference to me.
> I'll try Debian and Freeswitch.
> Thank you.
>
> 2012/8/29 Ken Rice <krice at freeswitch.org>
>
>> I used to use Centos6 and due to performance issues went to Debian 6 and
>> havent looked back...
>>
>> at some point we really need to get some different people to oprofile
>> their freeswitch under their specific configs and give us the report logs
>> so we can see why centos6 is behaving so badly...
>>
>> On Wed, Aug 29, 2012 at 7:41 AM, Roberto Linck <
>> roberto at i360tecnologia.com.br> wrote:
>>
>>> Hello everybody,
>>>
>>> I would like to know wich version of Freeswitch is the most stable one,
>>> in your opinion and wich linux distro is the best choice to use with this
>>> version.
>>> I make this question because I need put a Freeswitch in production to
>>> handle about 1500 concurrent calls. I know have the appropriate hardware,
>>> but I really don't know the most indicated version and linux distro. I'm
>>> using CentOS 6.2 but the newest version  of Freeswitch(1.2) doesn't work ok
>>> with CentOS 6.x, so I'm opened to any option, since this can be stable.
>>>
>>> I'll appreciate any contribution.
>>>
>>>
>>>
>>> Roberto Linck
>>> roberto at i360tecnologia.com.br
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 vbvbrj at gmail.com  Thu Aug 30 09:25:41 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Thu, 30 Aug 2012 08:25:41 +0300
Subject: [Freeswitch-users] uuid_broadcast muxing.
In-Reply-To: <CAKbxfG9CsU4U+xEYmVBWrGdXDDdsSfuXQEwQW7HP5bJQtogLGA@mail.gmail.com>
References: <503E39C6.40306@gmail.com>
	<CAKbxfG9CsU4U+xEYmVBWrGdXDDdsSfuXQEwQW7HP5bJQtogLGA@mail.gmail.com>
Message-ID: <503EF955.1080400@gmail.com>

On 30.08.2012 02:21, Anthony Minessale wrote:
> The error is harmless warning to remind you the local_stream is being ignored.
>
>
> On Wed, Aug 29, 2012 at 10:48 AM, Vbvbrj <vbvbrj at gmail.com> wrote:
>> Hello. I use uuid_broadcast to play a file to some leg:
>> api:executeString("uuid_broadcast "..caller_uuid.."
>> phrase::queue-position,"..pos.." aleg")
>> But this pauses the moh file which is played to member waiting in
>> callcenter. After two broadcasts, there is an error about leaking stream
>> and adjusting of moh play. Is there a command to do the same but mux
>> this playback of position with moh playing?

Yes. But a caller hears music jump, which is not very interesting to let 
the users hear that, as he/she may think that the pbx is broken.

Anyway, this is a suggestion that some one will be willing to implement. 
For now I use it as it is.

Thank you.



From kees at mroffice.org  Thu Aug 30 10:11:41 2012
From: kees at mroffice.org (Kees Varekamp)
Date: Thu, 30 Aug 2012 18:11:41 +1200
Subject: [Freeswitch-users] voicemaill
Message-ID: <CAHbVE90sDrU_=+8knWq04Q6s1-U6seZTkGj4pYDDCfWcKwE54A@mail.gmail.com>

I'm trying to get voicemail running on fs1.2 but i keep running into this:

2012-08-30 18:05:27.992966 [ERR] switch_xml.c:2986 Can't find macros tag.

Ive got this:
  <section name="languages" description="Language Management">
...
    <X-PRE-PROCESS cmd="include" data="lang/en/*.xml"/>

and in lang/en there is en.xml, which has this:
<include>
  <language name="en" say-module="en"
sound-prefix="$${sounds_dir}/en/us/callie" tts-engine="cepstral"
tts-voice="callie">
    <phrases>
      <macros>
        <X-PRE-PROCESS cmd="include" data="demo/*.xml"/> <!-- Note: this
now grabs whole subdir, previously grabbed only demo.xml -->
        <!--voicemail_en_tts is purely implemented with tts, we have the
files based one that is the default. -->
        <X-PRE-PROCESS cmd="include" data="vm/sounds.xml"/>  <!--
vm/tts.xml if you want to use tts and have cepstral -->
        <X-PRE-PROCESS cmd="include" data="dir/sounds.xml"/>  <!--
dir/tts.xml if you want to use tts and have cepstral -->
<X-PRE-PROCESS cmd="include" data="ivr/*.xml"/>  <!-- IVR and custom
phrases go here -->
      </macros>
    </phrases>
  </language>
</include>

Is my config wrong?

thanks,

Kees
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From miha at softnet.si  Thu Aug 30 10:27:11 2012
From: miha at softnet.si (Miha)
Date: Thu, 30 Aug 2012 08:27:11 +0200
Subject: [Freeswitch-users] group call pickup
In-Reply-To: <503E33C2.2090007@softnet.si>
References: <503E18D5.2020907@softnet.si>
	<CAC-LwPN4brMLWT5rxpUG--hiwUZz687SPq7C90yVf1BRZPQUJQ@mail.gmail.com>
	<503E2AF1.3030607@softnet.si>
	<CAC-LwPNHdfPwjpP5pkOOmvC7qWLuY4L288ca+LsqK9hZ5VvCyg@mail.gmail.com>
	<503E33C2.2090007@softnet.si>
Message-ID: <503F07BF.7070205@softnet.si>

Hi,

I also tried like it is written on wiki, with numberic groups. But still 
group pickup/call intercept do not work for me.

action application="bridge" 
data="[leg_timeout=30]user/${sip_to_user}.enterprise at xxx.xxx.xxx.xxx|user/${call_forwarding_number__nore}.enterprise at xxx.xxx.xxx.xxx, 
pickup/1"/>

<extension name="group_pickup">
   <condition field="destination_number" expression="^\*57(\d+)$">
     <action application="pickup" data="$1"/>
   </condition>
</extension>

2012-08-30 10:16:56.189010 [INFO] mod_dialplan_xml.c:485 Processing 018108753 <018108753.enterprise>->*571 in context xxx.xxx.xxx.xxx
Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx parsing [xxx.xxx.xxx.xxx->group-intercept-number] continue=false
Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Regex (FAIL) [group-intercept-number] destination_number(*571) =~ /^\*6(\d+)$/ break=on-false
Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx parsing [xxx.xxx.xxx.xxx->group-intercept] continue=false
Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Regex (PASS) [group-intercept] destination_number(*571) =~ /^\*57(\d+)$/ break=on-false
Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Action pickup(1)
2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State Change CS_ROUTING -> CS_EXECUTE
2012-08-30 10:16:56.189010 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx [BREAK]
2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State ROUTING going to sleep
2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) Running State Change CS_EXECUTE
2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State EXECUTE
2012-08-30 10:16:56.189010 [DEBUG] mod_sofia.c:241 sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx SOFIA EXECUTE
2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:196 sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Standard EXECUTE
EXECUTE sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx pickup(1)





On 8/29/2012 5:22 PM, Miha wrote:
> @Vik,
>
> sorry for my stupid questions...
>
> It is matched. As I am trying with account that are in the same callgroup.
>
> log:
>
> EXECUTE
> sofia/internal/018108754.enterprise at xxx.xxx.xxx.xxxbridge([leg_timeout=30]user/018108755.enterprise at xxx.xxx.xxx.xxx|user/.enterprise at xxx.xxx.xxx.xxx,
> pickup/test)
>
> afer *5:
>
> Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Action
> pickup(${callgroup})
>
> EXECUTE sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx pickup(test)
>
> thanks!
>
> Miha
>
> On 8/29/2012 5:02 PM, Vik Killa wrote:
>> pickup/XXXX only needs to match here
>> <action application="pickup" data="XXXX"/>
>>
>> You can use whatever variable as long as those two match.
>>
>> On Wed, Aug 29, 2012 at 10:45 AM, Miha <miha at softnet.si> wrote:
>>> Hi @Vik,
>>>
>>> I am having problem with understanding how this works. After I bridge
>>> data I put like this:
>>>
>>>     <anti-action application="bridge"
>>> data="[leg_timeout=30]user/${sip_to_user}.enterprise at xxx.xxx.xxx.xxx|user/${call_forwarding_number__nore}.enterprise at xxx.xxx.xxx.xxx,
>>> pickup/${callgroup
>>> }"/>
>>>
>>> To do pickup I am doing:
>>>
>>>
>>> <extension name="group-intercept">
>>>          <condition field="destination_number" expression="^\*5$">
>>> <action application="pickup" data="${groupcall}"/>
>>>             </condition>
>>>        </extension>
>>>
>>>
>>> What I am doing wrong or what I am missing?
>>>
>>> Thanks!
>>>
>>> Miha
>>> On 8/29/2012 3:49 PM, Vik Killa wrote:
>>>> That method of call pickup is out-dated, try using this, it works great:
>>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pickup
>>>>
>>>> On Wed, Aug 29, 2012 at 9:27 AM, Miha <miha at softnet.si> wrote:
>>>>> Hi,
>>>>>
>>>>> I have implemented call pickup, which is not working as it should.
>>>>>
>>>>> When call comes in, user from same group can pick it up, but problem appears
>>>>> in this scenario:
>>>>>
>>>>> A calls B. C do call pickup with *5 (in dialplan). OK this works (A is
>>>>> talking with C). But when D is calling E, if someone from same group pick
>>>>> this call, it do not pick D, it picks call A. Why? What I am doing wrong?
>>>>>
>>>>> I hope you understand what I mean:D
>>>>>
>>>>> my dialplan:
>>>>>
>>>>>     <action application="set"
>>>>> data="called_party_callgroup=${user_data(${destination_number}.enterprise at xxx.xxx.xxx.xxx
>>>>> var callgroup)}"/>
>>>>>                    <action application="hash"
>>>>> data="insert/${destination_number}/${called_party_callgroup}/${uuid}"/>
>>>>>                    <action application="hash"
>>>>> data="insert/last_dial/${called_party_callgroup}/${uuid}"/>
>>>>>
>>>>>
>>>>> intercept:
>>>>>
>>>>>
>>>>>           <extension name="group-intercept">
>>>>>          <condition field="destination_number" expression="^\*5$">
>>>>>            <action application="answer"/>
>>>>>             <action application="intercept"
>>>>> data="${hash(select/last_dial/${callgroup})}"/>
>>>>>
>>>>>            <action application="sleep" data="2000"/>
>>>>>          </condition>
>>>>>        </extension>
>>>>>
>>>>> Thanks!
>>>>> Miha
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 gvvsubhashkumar at gmail.com  Thu Aug 30 10:51:46 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Wed, 29 Aug 2012 23:51:46 -0700
Subject: [Freeswitch-users] How to write an Application
	tocontrolfreeSWITCH
In-Reply-To: <B4232C1F-C9F8-4188-B7C3-1DF6BC5A4511@gmail.com>
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
	<CALn233py3x9eJgXAvsE6q4dL85i4ho7cJU9aF9y5Z2sit1Q_TQ@mail.gmail.com>
	<4BA9BAACFB75451B85C50AF41025A1DE@coralocs.com>
	<CAFvodxW8R5OCOC8YxF24dCdfzrPVHubCx4ApPFnai_wJH+OPRw@mail.gmail.com>
	<6939CF2F4EC941D5A253C207A8F8CF2C@coralocs.com>
	<B4232C1F-C9F8-4188-B7C3-1DF6BC5A4511@gmail.com>
Message-ID: <CAFvodxVM+MnAeP9PZvmq_hrEooZNRZ4d33D4ae_qhUC+1TgLWw@mail.gmail.com>

Hi,

Can i get document how to use ESL and write  an application using it


Thanks,
Subhash.


On Tue, Aug 28, 2012 at 4:29 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello,
>
> I didt all that with ESL. To me there's nothing like it to do everything
> you want exactly as you want it. For the billing part, you will probably
> need to write your own stuff based on cdrs (maybe posted to a different
> server via curl_xml)
>
> You indeed need to read the books and start playing with FS, don't expect
> to have a complete solution in 5 minutes, unless you buy something, you
> need to GET TO KNOW FS inside out.
>
> Have fun! I know I did!
>
> David
>
> ---
>
> David Villasmil
>
>
> On Aug 28, 2012, at 9:04, "Sharad Garg" <sharad at coraltele.com> wrote:
>
> My comments are in red.
>
> regards
> Sharad
>
> ----- Original Message -----
> *From:* Subhash <gvvsubhashkumar at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 28, 2012 11:38 AM
> *Subject:* Re: [Freeswitch-users] How to write an Application
> tocontrolfreeSWITCH
>
>  Hi Sharad,
>
> We are trying to achieve the follwing things from our app.
>
>  Configuring FreeSWITCH
>
> >>>> Configuration of FS deponds upon what exactly your requirements are.
> Deponding upon that, desired config files will be configured. So it is
> little vast...not easy to comment on this.
>
>  Watching FreeSWITCH to see what calls/activity are currently taking place
>
> >>>>>> There are FS CLI commands available, which can be given either
> on FS CLI or through telnet (if you want to monitor the FS from your own
> app). These commands will let you know the running calls / call durations /
> call originators / receivers / used codecs, etc, etc. I think this will be
> more than enough for you.
>
>   Bridging and transfer.
>
> >>>>> Are you using FS as call server too ? if yes, just go through the
> wiki, you will get everything to bridge / transfer the calls.
>
>  Act as like an IVR.
>
>  >>>>> This is really something interesting. For use the FS as IVR
> server, FS really is a great and robust plateform. To provide more
> flexibility to your IVR, you can write your own Javascripts. When a call is
> answered by FS through dialpla, let the call handed over to JS. Now write
> JS to play the IVR. This is what we generally do.
>
> if you need some more help on this, give us the flow of your IVR, we will
> suggest the desired dialplan settings & sample JS.
>
> Thanks,
> Subhash.
> Regards
> Sharad
>
> On Mon, Aug 27, 2012 at 10:11 PM, Sharad Garg <sharad at coraltele.com>wrote:
>
>> Hello Subhash
>>
>> If you may elobrate little more, I think we will be able to help you out.
>>
>> Just explain what you want to achieve.
>>
>> Best Regards
>> Sharad
>>
>>
>>
>> ----- Original Message -----
>> From: "Mitch Capper" <mitch.capper at gmail.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, August 27, 2012 9:23 PM
>> Subject: Re: [Freeswitch-users] How to write an Application to
>> controlfreeSWITCH
>>
>>
>>  > look at the wiki for esl probably the easiest way if you need to you
>> > can also write fs modules in c++.
>> > ~mitch
>> >
>> > On Mon, Aug 27, 2012 at 8:19 AM, Subhash <gvvsubhashkumar at gmail.com>
>> > wrote:
>> >>
>> >> Hi All,
>> >>
>> >>        I am new to freeSWITCH platform.
>> >>
>> >>       I want to write an application using c++ to control freeSWITCH so
>> >> please guide me.
>> >>
>> >>       Thanks in advance.
>> >>
>> >> Thanks,
>> >> Subhash.
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.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
>> >>
>> >
>> >
>> >
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eagle.antonio at gmail.com  Thu Aug 30 10:55:07 2012
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Thu, 30 Aug 2012 06:55:07 +0000
Subject: [Freeswitch-users] How to write an Application
	tocontrolfreeSWITCH
In-Reply-To: <CAFvodxVM+MnAeP9PZvmq_hrEooZNRZ4d33D4ae_qhUC+1TgLWw@mail.gmail.com>
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
	<CALn233py3x9eJgXAvsE6q4dL85i4ho7cJU9aF9y5Z2sit1Q_TQ@mail.gmail.com>
	<4BA9BAACFB75451B85C50AF41025A1DE@coralocs.com>
	<CAFvodxW8R5OCOC8YxF24dCdfzrPVHubCx4ApPFnai_wJH+OPRw@mail.gmail.com>
	<6939CF2F4EC941D5A253C207A8F8CF2C@coralocs.com>
	<B4232C1F-C9F8-4188-B7C3-1DF6BC5A4511@gmail.com>
	<CAFvodxVM+MnAeP9PZvmq_hrEooZNRZ4d33D4ae_qhUC+1TgLWw@mail.gmail.com>
Message-ID: <CAF+3j-txnaxi9TgGEGbJz-MeFi8=cz9=w-bcJPhuFcdSDAJWuw@mail.gmail.com>

Search The Wiki
http://wiki.freeswitch.org/wiki/Main_Page
By ESL and you will get
http://wiki.freeswitch.org/wiki/ESL

examples at the end of the page

2012/8/30 Subhash <gvvsubhashkumar at gmail.com>

> Hi,
>
> Can i get document how to use ESL and write  an application using it
>
>
> Thanks,
> Subhash.
>
>
> On Tue, Aug 28, 2012 at 4:29 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello,
>>
>> I didt all that with ESL. To me there's nothing like it to do everything
>> you want exactly as you want it. For the billing part, you will probably
>> need to write your own stuff based on cdrs (maybe posted to a different
>> server via curl_xml)
>>
>> You indeed need to read the books and start playing with FS, don't expect
>> to have a complete solution in 5 minutes, unless you buy something, you
>> need to GET TO KNOW FS inside out.
>>
>> Have fun! I know I did!
>>
>> David
>>
>> ---
>>
>> David Villasmil
>>
>>
>> On Aug 28, 2012, at 9:04, "Sharad Garg" <sharad at coraltele.com> wrote:
>>
>> My comments are in red.
>>
>> regards
>> Sharad
>>
>> ----- Original Message -----
>> *From:* Subhash <gvvsubhashkumar at gmail.com>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Tuesday, August 28, 2012 11:38 AM
>> *Subject:* Re: [Freeswitch-users] How to write an Application
>> tocontrolfreeSWITCH
>>
>>  Hi Sharad,
>>
>> We are trying to achieve the follwing things from our app.
>>
>>  Configuring FreeSWITCH
>>
>> >>>> Configuration of FS deponds upon what exactly your requirements are.
>> Deponding upon that, desired config files will be configured. So it is
>> little vast...not easy to comment on this.
>>
>>  Watching FreeSWITCH to see what calls/activity are currently taking place
>>
>> >>>>>> There are FS CLI commands available, which can be given either
>> on FS CLI or through telnet (if you want to monitor the FS from your own
>> app). These commands will let you know the running calls / call durations /
>> call originators / receivers / used codecs, etc, etc. I think this will be
>> more than enough for you.
>>
>>   Bridging and transfer.
>>
>> >>>>> Are you using FS as call server too ? if yes, just go through the
>> wiki, you will get everything to bridge / transfer the calls.
>>
>>  Act as like an IVR.
>>
>>  >>>>> This is really something interesting. For use the FS as IVR
>> server, FS really is a great and robust plateform. To provide more
>> flexibility to your IVR, you can write your own Javascripts. When a call is
>> answered by FS through dialpla, let the call handed over to JS. Now write
>> JS to play the IVR. This is what we generally do.
>>
>> if you need some more help on this, give us the flow of your IVR, we will
>> suggest the desired dialplan settings & sample JS.
>>
>> Thanks,
>> Subhash.
>> Regards
>> Sharad
>>
>> On Mon, Aug 27, 2012 at 10:11 PM, Sharad Garg <sharad at coraltele.com>wrote:
>>
>>> Hello Subhash
>>>
>>> If you may elobrate little more, I think we will be able to help you out.
>>>
>>> Just explain what you want to achieve.
>>>
>>> Best Regards
>>> Sharad
>>>
>>>
>>>
>>> ----- Original Message -----
>>> From: "Mitch Capper" <mitch.capper at gmail.com>
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, August 27, 2012 9:23 PM
>>> Subject: Re: [Freeswitch-users] How to write an Application to
>>> controlfreeSWITCH
>>>
>>>
>>>  > look at the wiki for esl probably the easiest way if you need to you
>>> > can also write fs modules in c++.
>>> > ~mitch
>>> >
>>> > On Mon, Aug 27, 2012 at 8:19 AM, Subhash <gvvsubhashkumar at gmail.com>
>>> > wrote:
>>> >>
>>> >> Hi All,
>>> >>
>>> >>        I am new to freeSWITCH platform.
>>> >>
>>> >>       I want to write an application using c++ to control freeSWITCH
>>> so
>>> >> please guide me.
>>> >>
>>> >>       Thanks in advance.
>>> >>
>>> >> Thanks,
>>> >> Subhash.
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> 
>>> >> 
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://wiki.freeswitch.org
>>> >> http://www.cluecon.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
>>> >>
>>> >
>>> >
>>> >
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mgg at giagnocavo.net  Thu Aug 30 11:03:59 2012
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Thu, 30 Aug 2012 07:03:59 +0000
Subject: [Freeswitch-users] Best way to stream record session?
Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FC202C@BLUPRD0711MB413.namprd07.prod.outlook.com>

Hi all,

	I want to record and stream a (bridged) call to a non-channel endpoint. Sorta like mod_spy, but targeting a TCP or UDP connection or even a one-way RTP stream. What's the best way to approach this? What about using normal session record to a file, then "streaming" that file from an external program? Is FS guaranteed to write to the wav file regularly (like every second or so) in a manner that another program can keep reading? Or should I specify a file like "/dev/udp/1.2.3.4/666"?

	End goal is to have a separate machine that can provide a service like Internet radio and have it tap into live calls.

Cheers,
Michael




From kees at mroffice.org  Thu Aug 30 11:04:03 2012
From: kees at mroffice.org (Kees Varekamp)
Date: Thu, 30 Aug 2012 19:04:03 +1200
Subject: [Freeswitch-users] voicemaill
In-Reply-To: <CAHbVE90sDrU_=+8knWq04Q6s1-U6seZTkGj4pYDDCfWcKwE54A@mail.gmail.com>
References: <CAHbVE90sDrU_=+8knWq04Q6s1-U6seZTkGj4pYDDCfWcKwE54A@mail.gmail.com>
Message-ID: <CAHbVE913UNSY=SAd2Q88VXvzQ4PfFJmm1MtS9c8YYGTGkq59yA@mail.gmail.com>

Fixed - was configuring the wrong server...



On Thu, Aug 30, 2012 at 6:11 PM, Kees Varekamp <kees at mroffice.org> wrote:

> I'm trying to get voicemail running on fs1.2 but i keep running into this:
>
> 2012-08-30 18:05:27.992966 [ERR] switch_xml.c:2986 Can't find macros tag.
>
> Ive got this:
>   <section name="languages" description="Language Management">
> ...
>     <X-PRE-PROCESS cmd="include" data="lang/en/*.xml"/>
>
> and in lang/en there is en.xml, which has this:
> <include>
>   <language name="en" say-module="en"
> sound-prefix="$${sounds_dir}/en/us/callie" tts-engine="cepstral"
> tts-voice="callie">
>     <phrases>
>       <macros>
>         <X-PRE-PROCESS cmd="include" data="demo/*.xml"/> <!-- Note: this
> now grabs whole subdir, previously grabbed only demo.xml -->
>         <!--voicemail_en_tts is purely implemented with tts, we have the
> files based one that is the default. -->
>         <X-PRE-PROCESS cmd="include" data="vm/sounds.xml"/>  <!--
> vm/tts.xml if you want to use tts and have cepstral -->
>         <X-PRE-PROCESS cmd="include" data="dir/sounds.xml"/>  <!--
> dir/tts.xml if you want to use tts and have cepstral -->
> <X-PRE-PROCESS cmd="include" data="ivr/*.xml"/>  <!-- IVR and custom
> phrases go here -->
>       </macros>
>     </phrases>
>   </language>
> </include>
>
> Is my config wrong?
>
> thanks,
>
> Kees
>
>


-- 
Kees Varekamp
*MROffice Director*
3097A Great North Road, 0600, Auckland, New Zealand
Phone: +6499743734
Skype: kees.mroffice
Web: mro <http://mro-global.com>ffice.org

*MROffice - Meeting Market Research Needs*
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From alex at jajah.com  Thu Aug 30 11:07:40 2012
From: alex at jajah.com (Alex Massover)
Date: Thu, 30 Aug 2012 10:07:40 +0300
Subject: [Freeswitch-users] Prevent A leg from hangup after bridge
	with	inbound ESL socket
In-Reply-To: <CAKzWOxW+jCoFfd-e0GkBDvsez4f66xqDLnzpYM6-QesJ+3Md7g@mail.gmail.com>
References: <569384504C492C4580E88B5D54DFEAEA30CAFD33B0@jjex01.jajah.dublin>
	<CAKzWOxW+jCoFfd-e0GkBDvsez4f66xqDLnzpYM6-QesJ+3Md7g@mail.gmail.com>
Message-ID: <569384504C492C4580E88B5D54DFEAEA30CAFD3530@jjex01.jajah.dublin>

Hi Michael,

Thanks, that works in scenario when B legs response with 100 and then let's say 486. But if B legs do ringing, i.e. 100, 180/183, 486 it doesn't work.

I found this in wiki "By the way, you'll be unable to rewrite the hangup cause for a bridge that gets a 180 or 183 packet from the gateway before getting a 4xx, 5xx or 6xx packet (because those bridges don't 'fail')."

I understand that continue_on_fail won't help with this scenario. I see that's a popular topic in the list, but nobody got a solution.

BR, Alex.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Wednesday, August 29, 2012 6:21 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Prevent A leg from hangup after bridge with inbound ESL socket

Try this:
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail

-MC
On Wed, Aug 29, 2012 at 6:57 AM, Alex Massover <alex at jajah.com<mailto:alex at jajah.com>> wrote:
Hi,

I have a very simple dialplan that just do park for incoming calls. All rest of leg management is done via ESL inbound socket.

I'm trying to do the same behavior like in this dialplan example, but from ESL inbound socket:
<action application="set" data="hangup_after_bridge=false"/>
   <action application="bridge" data="sofia/internal/1001@$${domain}"/>
   <action application="bridge" data="sofia/internal/1002@$${domain}"/>

The problem is with bridge API, if B leg doesn't answer (e.g. 404, or busy), A leg disconnects. But I'm trying to prevent A leg from disconnecting in order to do bridge to other place.

Looks like hangup_after_bridge=false, park_after_bridge=true, transfer_after_bridge etc don't have any effect when bridge done from inbound socket. A leg disconnects always.

Is there any way to keep A leg after bridge with inbound socket? I'm aware of originate, but prefer to user bridge.




--
Best Regards,
Alex Massover


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org

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From freeswitch-list at puzzled.xs4all.nl  Thu Aug 30 16:02:40 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 30 Aug 2012 14:02:40 +0200
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <CAPcNkdefcnY+_By3y1nUS8H=9R48jq4m2fmnqaDc=5sNCK2xUw@mail.gmail.com>
References: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
	<CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
	<CAFTa0ycv_pTjCfXL7tYT5vcx=jad68_r+tcRdnC84QHXnr4nVA@mail.gmail.com>
	<CAPcNkdefcnY+_By3y1nUS8H=9R48jq4m2fmnqaDc=5sNCK2xUw@mail.gmail.com>
Message-ID: <503F5660.4030304@puzzled.xs4all.nl>

On 08/30/2012 05:17 AM, Ken Rice wrote:
> chances are, pretty much whatever platform you pick will work unless you
> are doing something really crazy (like trying to cram an OC12 worth of
> calls through 1 machine
>
> (why you would want to risk that much business on one machine I am not
> sure, but I am sure someone somewhere will try it eventually)

That's about 7000 calls. Might make for a nice proof of concept of how 
scalable FreeSWITCH is. If someone has an itch then here's a link to a 
few OC12 cards and OC12 analysis & simulation software:

http://www.gl.com/OC3-OC12-analysis-emulation-card.html
http://oem.imagestream.com/PCI_1000.html
http://xalyo.com/products/detail/xs-3100

Pictures and video or it did not exist :)

Regards,
Patrick



From sdame at 207me.com  Thu Aug 30 16:10:34 2012
From: sdame at 207me.com (Stephen Dame)
Date: Thu, 30 Aug 2012 08:10:34 -0400
Subject: [Freeswitch-users] Best way to stream record session?
In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FC202C@BLUPRD0711MB413.namprd07.prod.outlook.com>
References: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FC202C@BLUPRD0711MB413.namprd07.prod.outlook.com>
Message-ID: <000001cd86a8$75efcd10$61cf6730$@207me.com>

Mod_shout  configured  to an icecast server works well for this.   The wiki
has examples.

The latest freeswitch cookbook, also has section on telecast? Where you
access call from http served directly from freeswitch.  Not sure how this
scales if you have bunch of listeners.

Regards,
Stephen

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Giagnocavo
Sent: Thursday, August 30, 2012 3:04 AM
To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)
Subject: [Freeswitch-users] Best way to stream record session?

Hi all,

	I want to record and stream a (bridged) call to a non-channel
endpoint. Sorta like mod_spy, but targeting a TCP or UDP connection or even
a one-way RTP stream. What's the best way to approach this? What about using
normal session record to a file, then "streaming" that file from an external
program? Is FS guaranteed to write to the wav file regularly (like every
second or so) in a manner that another program can keep reading? Or should I
specify a file like "/dev/udp/1.2.3.4/666"?

	End goal is to have a separate machine that can provide a service
like Internet radio and have it tap into live calls.

Cheers,
Michael



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 vipkilla at gmail.com  Thu Aug 30 16:13:06 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Thu, 30 Aug 2012 08:13:06 -0400
Subject: [Freeswitch-users] group call pickup
In-Reply-To: <503F07BF.7070205@softnet.si>
References: <503E18D5.2020907@softnet.si>
	<CAC-LwPN4brMLWT5rxpUG--hiwUZz687SPq7C90yVf1BRZPQUJQ@mail.gmail.com>
	<503E2AF1.3030607@softnet.si>
	<CAC-LwPNHdfPwjpP5pkOOmvC7qWLuY4L288ca+LsqK9hZ5VvCyg@mail.gmail.com>
	<503E33C2.2090007@softnet.si> <503F07BF.7070205@softnet.si>
Message-ID: <CAC-LwPOH_5nbbZ3Z6iM3JdkFrmZjP_NMBCjV7AB1tHux2GzhkA@mail.gmail.com>

I'm not sure where the problem is here... try doing something simple like this:
<action application="bridge"
data="leg_timeout=30]user/${sip_to_user}.enterprise at xxx.xxx.xxx.xxx|user/${call_forwarding_number__nore}.enterprise at xxx.xxx.xxx.xxx,pickup/1"/>

and this:
<extension name="group_pickup">
   <condition field="destination_number" expression="^\*57(\d+)$">
     <action application="pickup" data="1"/>
   </condition>
</extension>



On Thu, Aug 30, 2012 at 2:27 AM, Miha <miha at softnet.si> wrote:
> Hi,
>
> I also tried like it is written on wiki, with numberic groups. But still
> group pickup/call intercept do not work for me.
>
> action application="bridge"
> data="[leg_timeout=30]user/${sip_to_user}.enterprise at xxx.xxx.xxx.xxx|user/${call_forwarding_number__nore}.enterprise at xxx.xxx.xxx.xxx,
> pickup/1"/>
>
> <extension name="group_pickup">
>    <condition field="destination_number" expression="^\*57(\d+)$">
>      <action application="pickup" data="$1"/>
>    </condition>
> </extension>
>
> 2012-08-30 10:16:56.189010 [INFO] mod_dialplan_xml.c:485 Processing 018108753 <018108753.enterprise>->*571 in context xxx.xxx.xxx.xxx
> Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx parsing [xxx.xxx.xxx.xxx->group-intercept-number] continue=false
> Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Regex (FAIL) [group-intercept-number] destination_number(*571) =~ /^\*6(\d+)$/ break=on-false
> Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx parsing [xxx.xxx.xxx.xxx->group-intercept] continue=false
> Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Regex (PASS) [group-intercept] destination_number(*571) =~ /^\*57(\d+)$/ break=on-false
> Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Action pickup(1)
> 2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State Change CS_ROUTING -> CS_EXECUTE
> 2012-08-30 10:16:56.189010 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx [BREAK]
> 2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State ROUTING going to sleep
> 2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) Running State Change CS_EXECUTE
> 2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State EXECUTE
> 2012-08-30 10:16:56.189010 [DEBUG] mod_sofia.c:241 sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx SOFIA EXECUTE
> 2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:196 sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Standard EXECUTE
> EXECUTE sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx pickup(1)
>
>
>
>
>
> On 8/29/2012 5:22 PM, Miha wrote:
>> @Vik,
>>
>> sorry for my stupid questions...
>>
>> It is matched. As I am trying with account that are in the same callgroup.
>>
>> log:
>>
>> EXECUTE
>> sofia/internal/018108754.enterprise at xxx.xxx.xxx.xxxbridge([leg_timeout=30]user/018108755.enterprise at xxx.xxx.xxx.xxx|user/.enterprise at xxx.xxx.xxx.xxx,
>> pickup/test)
>>
>> afer *5:
>>
>> Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Action
>> pickup(${callgroup})
>>
>> EXECUTE sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx pickup(test)
>>
>> thanks!
>>
>> Miha
>>
>> On 8/29/2012 5:02 PM, Vik Killa wrote:
>>> pickup/XXXX only needs to match here
>>> <action application="pickup" data="XXXX"/>
>>>
>>> You can use whatever variable as long as those two match.
>>>
>>> On Wed, Aug 29, 2012 at 10:45 AM, Miha <miha at softnet.si> wrote:
>>>> Hi @Vik,
>>>>
>>>> I am having problem with understanding how this works. After I bridge
>>>> data I put like this:
>>>>
>>>>     <anti-action application="bridge"
>>>> data="[leg_timeout=30]user/${sip_to_user}.enterprise at xxx.xxx.xxx.xxx|user/${call_forwarding_number__nore}.enterprise at xxx.xxx.xxx.xxx,
>>>> pickup/${callgroup
>>>> }"/>
>>>>
>>>> To do pickup I am doing:
>>>>
>>>>
>>>> <extension name="group-intercept">
>>>>          <condition field="destination_number" expression="^\*5$">
>>>> <action application="pickup" data="${groupcall}"/>
>>>>             </condition>
>>>>        </extension>
>>>>
>>>>
>>>> What I am doing wrong or what I am missing?
>>>>
>>>> Thanks!
>>>>
>>>> Miha
>>>> On 8/29/2012 3:49 PM, Vik Killa wrote:
>>>>> That method of call pickup is out-dated, try using this, it works great:
>>>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pickup
>>>>>
>>>>> On Wed, Aug 29, 2012 at 9:27 AM, Miha <miha at softnet.si> wrote:
>>>>>> Hi,
>>>>>>
>>>>>> I have implemented call pickup, which is not working as it should.
>>>>>>
>>>>>> When call comes in, user from same group can pick it up, but problem appears
>>>>>> in this scenario:
>>>>>>
>>>>>> A calls B. C do call pickup with *5 (in dialplan). OK this works (A is
>>>>>> talking with C). But when D is calling E, if someone from same group pick
>>>>>> this call, it do not pick D, it picks call A. Why? What I am doing wrong?
>>>>>>
>>>>>> I hope you understand what I mean:D
>>>>>>
>>>>>> my dialplan:
>>>>>>
>>>>>>     <action application="set"
>>>>>> data="called_party_callgroup=${user_data(${destination_number}.enterprise at xxx.xxx.xxx.xxx
>>>>>> var callgroup)}"/>
>>>>>>                    <action application="hash"
>>>>>> data="insert/${destination_number}/${called_party_callgroup}/${uuid}"/>
>>>>>>                    <action application="hash"
>>>>>> data="insert/last_dial/${called_party_callgroup}/${uuid}"/>
>>>>>>
>>>>>>
>>>>>> intercept:
>>>>>>
>>>>>>
>>>>>>           <extension name="group-intercept">
>>>>>>          <condition field="destination_number" expression="^\*5$">
>>>>>>            <action application="answer"/>
>>>>>>             <action application="intercept"
>>>>>> data="${hash(select/last_dial/${callgroup})}"/>
>>>>>>
>>>>>>            <action application="sleep" data="2000"/>
>>>>>>          </condition>
>>>>>>        </extension>
>>>>>>
>>>>>> Thanks!
>>>>>> Miha
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 krice at freeswitch.org  Thu Aug 30 16:13:58 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 30 Aug 2012 07:13:58 -0500
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <503F5660.4030304@puzzled.xs4all.nl>
References: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
	<CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
	<CAFTa0ycv_pTjCfXL7tYT5vcx=jad68_r+tcRdnC84QHXnr4nVA@mail.gmail.com>
	<CAPcNkdefcnY+_By3y1nUS8H=9R48jq4m2fmnqaDc=5sNCK2xUw@mail.gmail.com>
	<503F5660.4030304@puzzled.xs4all.nl>
Message-ID: <CAPcNkddziUvgSf=0Pv=UDpGSJoLTftitVWiOREedQQ1ppjdpdA@mail.gmail.com>

lol... actually using a pretty heavy duty machine (i think it was like a
 quad hexa-core machine) Anthony demonstrated 10K concurrent call playing a
30 second sound file,  while initiating 1000 new calls per second at
cluecon in 2011...

so the scalability is there at least in some form.... the problem is, do
you really want 5000 concurrent calls potentially $100/minute worth of
cheap calls going through a machine that can hard fail and leave you losing
5K to 10K CDRs...

K

On Thu, Aug 30, 2012 at 7:02 AM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 08/30/2012 05:17 AM, Ken Rice wrote:
> > chances are, pretty much whatever platform you pick will work unless you
> > are doing something really crazy (like trying to cram an OC12 worth of
> > calls through 1 machine
> >
> > (why you would want to risk that much business on one machine I am not
> > sure, but I am sure someone somewhere will try it eventually)
>
> That's about 7000 calls. Might make for a nice proof of concept of how
> scalable FreeSWITCH is. If someone has an itch then here's a link to a
> few OC12 cards and OC12 analysis & simulation software:
>
> http://www.gl.com/OC3-OC12-analysis-emulation-card.html
> http://oem.imagestream.com/PCI_1000.html
> http://xalyo.com/products/detail/xs-3100
>
> Pictures and video or it did not exist :)
>
> Regards,
> Patrick
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-list at puzzled.xs4all.nl  Thu Aug 30 16:41:46 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 30 Aug 2012 14:41:46 +0200
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <CAPcNkddziUvgSf=0Pv=UDpGSJoLTftitVWiOREedQQ1ppjdpdA@mail.gmail.com>
References: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
	<CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
	<CAFTa0ycv_pTjCfXL7tYT5vcx=jad68_r+tcRdnC84QHXnr4nVA@mail.gmail.com>
	<CAPcNkdefcnY+_By3y1nUS8H=9R48jq4m2fmnqaDc=5sNCK2xUw@mail.gmail.com>
	<503F5660.4030304@puzzled.xs4all.nl>
	<CAPcNkddziUvgSf=0Pv=UDpGSJoLTftitVWiOREedQQ1ppjdpdA@mail.gmail.com>
Message-ID: <503F5F8A.8060103@puzzled.xs4all.nl>

On 08/30/2012 02:13 PM, Ken Rice wrote:
> lol... actually using a pretty heavy duty machine (i think it was like a
>   quad hexa-core machine) Anthony demonstrated 10K concurrent call
> playing a 30 second sound file,  while initiating 1000 new calls per
> second at cluecon in 2011...

I remember mentioning similar numbers at a FreeSWITCH presentation in 
the Netherlands a few years back (but only after verifying the numbers 
several times with Anthony and Brian since they were quite hefty). At 
the presentation oej was in the audience and when I mentioned those 
numbers he jumped up and said "impossible!". That was a fun discussion. 
I referred him to Anthony but never heard if he actually did and if he 
accepted those numbers to be true.

> so the scalability is there at least in some form.... the problem is, do
> you really want 5000 concurrent calls potentially $100/minute worth of
> cheap calls going through a machine that can hard fail and leave you
> losing 5K to 10K CDRs...

I would not. And I prefer Gigabit Ethernet over an OC12 link any day. 
But if you use something like a Stratus box then it should be possible 
to get at least 5 nines availability. The question is off course if such 
a solution is financially viable as those Stratus boxes don't come cheap.

Regards,
Patrick



From daveh at beachdognet.com  Thu Aug 30 16:44:02 2012
From: daveh at beachdognet.com (Dave Horton)
Date: Thu, 30 Aug 2012 08:44:02 -0400
Subject: [Freeswitch-users] how to play a prompt after connection
References: <23F564DC-911A-4D5B-A63F-D420C755B216@beachdognet.com>
	<F62AB8C5-4184-4937-9AE0-27EF32F000ED@beachdognet.com>
Message-ID: <k1nn6g$kd6$1@ger.gmane.org>

I did try that suggestion (api_on_answer uuidbroadcast) and it worked 
fine if I played it to the a leg, but when I tried playing to the b leg 
the prompt was incredibly choppy and slow, and delayed the cutthru of a 
and b legs by a long time.  Interestingly, the prompt doesn't even 
start playing to the b party until some RTP is received from b for some 
reason, and then it sort of slowly chuffs out the prompt in small 
choppy chunks.

I then tried Michael's follow on suggestion of using 
bridge_pre_execute_bleg_app, but that didn't work at all -- no prompt 
heard.

I then decided to try the outdial as a bridged conference, and then 
play a prompt to the conference after the b party picks up.  This may 
actually be the best option for me as I need the prompt to be saved in 
the recording of the call, and I am guessing when I stream it out on a 
bridged call it may not show up in the recording.

However, I had no luck getting a bridged conference outdial to work at 
all -- the outdial does work successfully, but a and b parties don't 
hear each other so one must not be in the conference.  And then when 
the b party hangs up the a party is not automatically hung up, which 
should happen in a bridged conference if I understand things correctly.

So, while I am still interested in what is causing the prompt on the 
bridge call to sound so bad when played to the b leg, perhaps what is 
more important to me now is getting the bridged conference to work.  
Here is what I am doing, in case anyone can point out the problem:

session:execute("set","progress_timeout=60") ;
session:execute("set","call_timeout=120") ;
session:setVariable("hangup_after_bridge","true") ;
session:execute("conference","bridge:_uuid_ at simple:sofia/normal_customer/${outdial}@${egress_gateway}") 
;
session:setVariable("conference_enter_sound",this_call_is_being_recorded) ;

(I created a conference profile called 'simple' that turns off caller 
controls; interestingly, that doesn't seem to be picked up either as 
the debug logging shows it attaching the default caller controls)

Dave

On 2012-08-29 11:24:00 +0000, Dave Horton said:

>> 
>> Set api_on_answer to execute
>> uuid_broadcast<
>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
>>> 
>> which
>> will let you pick which legs to play the recording to.
>> You can pass in the UUID with ${uuid}
>> 
>> -Avi
>> 
> I tried this, and it worked.....sort of.  If I play the prompt the A 
> leg it is fine -- caller hears the prompt, then is connected to called 
> party.  If I play it to the B leg, however, the prompt does not seem to 
> start playing at all until the B party actually starts talking, and 
> then the prompt is played extremely choppy and slow.....it takes about 
> 10 seconds to play a 4 second prompt.  Meanwhile, the caller and called 
> party can't hear each other.  Any idea what is going on and how to fix 
> it?  I am doing it in lua as follows:
> 
> session:execute("export","nolocal:jitterbuffer_msec=100") ;
> session:setVariable("RECORD_STEREO","true") ;
> session:setVariable("aleg_uuid",uuid) ;
> session:execute("record_session", outfile) ;
> session:execute("set","progress_timeout=60") ;
> session:execute("set","call_timeout=120") ;
> session:setVariable("hangup_after_bridge","true") ;
> session:execute("bridge","sofia/normal_customer/${outdial}@${egress_gateway}") 
> ;
> --session:execute("bridge","{api_on_answer='uuid_broadcast " .. uuid .. 
> " " .. this_call_is_being_recorded .. " " .. 
> "bleg'}sofia/normal_customer/${outdial}@${egress_gateway}") ;
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 krice at freeswitch.org  Thu Aug 30 17:14:23 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 30 Aug 2012 08:14:23 -0500
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <503F5F8A.8060103@puzzled.xs4all.nl>
References: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
	<CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
	<CAFTa0ycv_pTjCfXL7tYT5vcx=jad68_r+tcRdnC84QHXnr4nVA@mail.gmail.com>
	<CAPcNkdefcnY+_By3y1nUS8H=9R48jq4m2fmnqaDc=5sNCK2xUw@mail.gmail.com>
	<503F5660.4030304@puzzled.xs4all.nl>
	<CAPcNkddziUvgSf=0Pv=UDpGSJoLTftitVWiOREedQQ1ppjdpdA@mail.gmail.com>
	<503F5F8A.8060103@puzzled.xs4all.nl>
Message-ID: <CAPcNkddSHTW+J361UEOMfWaAOW8xC8AWdFLzBrNSrBuAjXy0SA@mail.gmail.com>

On Thu, Aug 30, 2012 at 7:41 AM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 08/30/2012 02:13 PM, Ken Rice wrote:
> > lol... actually using a pretty heavy duty machine (i think it was like a
> >   quad hexa-core machine) Anthony demonstrated 10K concurrent call
> > playing a 30 second sound file,  while initiating 1000 new calls per
> > second at cluecon in 2011...
>
> I remember mentioning similar numbers at a FreeSWITCH presentation in
> the Netherlands a few years back (but only after verifying the numbers
> several times with Anthony and Brian since they were quite hefty). At
> the presentation oej was in the audience and when I mentioned those
> numbers he jumped up and said "impossible!". That was a fun discussion.
> I referred him to Anthony but never heard if he actually did and if he
> accepted those numbers to be true.
>
>
I actually witnessed his demonstration at cluecon where that happened...

and I have personally demonstrated 2500 concurrent media playback calls
running FreeSWITCH on a Dell SC420 playing a sound file from ram drive...
so i know its possible to reach much higher density on newer hardware....



> > so the scalability is there at least in some form.... the problem is, do
> > you really want 5000 concurrent calls potentially $100/minute worth of
> > cheap calls going through a machine that can hard fail and leave you
> > losing 5K to 10K CDRs...
>
> I would not. And I prefer Gigabit Ethernet over an OC12 link any day.
> But if you use something like a Stratus box then it should be possible
> to get at least 5 nines availability. The question is off course if such
> a solution is financially viable as those Stratus boxes don't come cheap.



The problem with Stratus would be not only are they very expensive, but
their configurations tend to not be the highest end CPUs/RAM configs
without a Fortune500 size budget...
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From lon at kickasspixels.com  Thu Aug 30 19:59:37 2012
From: lon at kickasspixels.com (Lon Baker)
Date: Thu, 30 Aug 2012 08:59:37 -0700
Subject: [Freeswitch-users] Eavesdrop with bind_digit_action
In-Reply-To: <mailman.17959.1346332495.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.17959.1346332495.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <19C8F06C-3FCD-4FBA-84E4-B656E7846479@kickasspixels.com>

I'm trying to add the ability to screen voicemail being left, and allow the callee to pull the call back from voicemail with a DTMF. I thought I could use eavesdrop and bind_digit_action, but its not working.

The bound DTMF is logged in the console, but the exec action is never triggered.

Caller is correctly sent to voicemail
Callee, who elected to eavesdrop, can hear the voicemail being left.
Pressing 9 for the bind_digit_action, show up in the console.

But, exec extension does not go off.

Any help would be appreciated.

--
Lon



From msc at freeswitch.org  Thu Aug 30 20:10:34 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 09:10:34 -0700
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAAdTMa-keR12qGRZOQdOijR8PAZU3GyGORcT7ABNnfoJbARdfA@mail.gmail.com>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
	<CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
	<CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
	<CAAdTMa-keR12qGRZOQdOijR8PAZU3GyGORcT7ABNnfoJbARdfA@mail.gmail.com>
Message-ID: <CAKzWOxUSB7OT78xHcKsXdxu7T-qwpdve4FtxPe5NpNTTAu3a-g@mail.gmail.com>

Reports are that when the load starts to increase there's an exponential
increase in kernel calls, causing all kinds of havoc. If you can run it in
oprofile and get the logs to Ken Rice that would be helpful.
-MC

On Wed, Aug 29, 2012 at 3:21 PM, BookBag <asaad2 at gmail.com> wrote:

> Woe is me. I'm using centos 6 and I figured that performance issues was
> due to my cheap box and was actually planning to buy a more powerful
> computer.
>
> What specific issues did you guys encounter and has anyone tried the new
> 6.3 version. I would like to keep an eye out for these problems on my
> distro.
> On Aug 29, 2012 5:10 PM, "Ken Rice" <krice at freeswitch.org> wrote:
>
>> Centos5 worked well for me, Centos6 worked for crap... I went to debian 6
>> (squeeze) and problems went a way... using the latest stable on the debian6
>> line installing from the netinstall (non-free variant as I have hardware
>> thats not supported under the main netinstall iso) and then installing
>> freeswitch via the instructions on the wiki and no problems so far
>>
>> (the centos6 issues were bad enough i moved my dev box to debian6)
>>
>> On Wed, Aug 29, 2012 at 4:01 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Wed, Aug 29, 2012 at 1:56 PM, Vik Killa <vipkilla at gmail.com> wrote:
>>>
>>>> What IS the recommended linux distro to use for optimum performance?
>>>> I've heard debian, can someone be more specific though, like what
>>>> version, kernel, etc... ?
>>>>
>>>
>>> We try to stay out of the OS Holy Wars, so we don't have an "official"
>>> recommended distro. However, I know more than a few people (disclosure: my
>>> self included) who have been using Debian Squeeze 32 and 64 bit with zero
>>> drama, zero issues, and solid performance.
>>>
>>> --
>>> Michael S Collins
>>> Twitter: @mercutioviz
>>> http://www.FreeSWITCH.org
>>> http://www.ClueCon.com
>>> http://www.OSTAG.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Aug 30 20:13:50 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 09:13:50 -0700
Subject: [Freeswitch-users] How to write an Application
	tocontrolfreeSWITCH
In-Reply-To: <CAF+3j-txnaxi9TgGEGbJz-MeFi8=cz9=w-bcJPhuFcdSDAJWuw@mail.gmail.com>
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
	<CALn233py3x9eJgXAvsE6q4dL85i4ho7cJU9aF9y5Z2sit1Q_TQ@mail.gmail.com>
	<4BA9BAACFB75451B85C50AF41025A1DE@coralocs.com>
	<CAFvodxW8R5OCOC8YxF24dCdfzrPVHubCx4ApPFnai_wJH+OPRw@mail.gmail.com>
	<6939CF2F4EC941D5A253C207A8F8CF2C@coralocs.com>
	<B4232C1F-C9F8-4188-B7C3-1DF6BC5A4511@gmail.com>
	<CAFvodxVM+MnAeP9PZvmq_hrEooZNRZ4d33D4ae_qhUC+1TgLWw@mail.gmail.com>
	<CAF+3j-txnaxi9TgGEGbJz-MeFi8=cz9=w-bcJPhuFcdSDAJWuw@mail.gmail.com>
Message-ID: <CAKzWOxXUCcFPpDo8G9p1rFu_YfohgOkxZqC+7S2rujxLdPQk-Q@mail.gmail.com>

Also, the two FS books have information on getting yourself familiar with
ESL.
FS 1.0.6 book (ch 9) -
https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book
FS cookbook (ch 4) - http://link.packtpub.com/nuIOlX

Also, depending on what language you are familiar with there are some
simple examples in src/libs/esl/xxx where xxx = perl, python, ruby, php,
tcl, lua, managed, etc.

-MC

On Wed, Aug 29, 2012 at 11:55 PM, Antonio Teixeira
<eagle.antonio at gmail.com>wrote:

> Search The Wiki
> http://wiki.freeswitch.org/wiki/Main_Page
> By ESL and you will get
> http://wiki.freeswitch.org/wiki/ESL
>
> examples at the end of the page
>
>
> 2012/8/30 Subhash <gvvsubhashkumar at gmail.com>
>
>> Hi,
>>
>> Can i get document how to use ESL and write  an application using it
>>
>>
>> Thanks,
>> Subhash.
>>
>>
>> On Tue, Aug 28, 2012 at 4:29 AM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> I didt all that with ESL. To me there's nothing like it to do everything
>>> you want exactly as you want it. For the billing part, you will probably
>>> need to write your own stuff based on cdrs (maybe posted to a different
>>> server via curl_xml)
>>>
>>> You indeed need to read the books and start playing with FS, don't
>>> expect to have a complete solution in 5 minutes, unless you buy something,
>>> you need to GET TO KNOW FS inside out.
>>>
>>> Have fun! I know I did!
>>>
>>> David
>>>
>>> ---
>>>
>>> David Villasmil
>>>
>>>
>>> On Aug 28, 2012, at 9:04, "Sharad Garg" <sharad at coraltele.com> wrote:
>>>
>>> My comments are in red.
>>>
>>> regards
>>> Sharad
>>>
>>> ----- Original Message -----
>>> *From:* Subhash <gvvsubhashkumar at gmail.com>
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Sent:* Tuesday, August 28, 2012 11:38 AM
>>> *Subject:* Re: [Freeswitch-users] How to write an Application
>>> tocontrolfreeSWITCH
>>>
>>>  Hi Sharad,
>>>
>>> We are trying to achieve the follwing things from our app.
>>>
>>>  Configuring FreeSWITCH
>>>
>>> >>>> Configuration of FS deponds upon what exactly your requirements
>>> are. Deponding upon that, desired config files will be configured. So it is
>>> little vast...not easy to comment on this.
>>>
>>>  Watching FreeSWITCH to see what calls/activity are currently taking
>>> place
>>>
>>> >>>>>> There are FS CLI commands available, which can be given either
>>> on FS CLI or through telnet (if you want to monitor the FS from your own
>>> app). These commands will let you know the running calls / call durations /
>>> call originators / receivers / used codecs, etc, etc. I think this will be
>>> more than enough for you.
>>>
>>>   Bridging and transfer.
>>>
>>> >>>>> Are you using FS as call server too ? if yes, just go through the
>>> wiki, you will get everything to bridge / transfer the calls.
>>>
>>>  Act as like an IVR.
>>>
>>>  >>>>> This is really something interesting. For use the FS as IVR
>>> server, FS really is a great and robust plateform. To provide more
>>> flexibility to your IVR, you can write your own Javascripts. When a call is
>>> answered by FS through dialpla, let the call handed over to JS. Now write
>>> JS to play the IVR. This is what we generally do.
>>>
>>> if you need some more help on this, give us the flow of your IVR, we
>>> will suggest the desired dialplan settings & sample JS.
>>>
>>> Thanks,
>>> Subhash.
>>> Regards
>>> Sharad
>>>
>>> On Mon, Aug 27, 2012 at 10:11 PM, Sharad Garg <sharad at coraltele.com>wrote:
>>>
>>>> Hello Subhash
>>>>
>>>> If you may elobrate little more, I think we will be able to help you
>>>> out.
>>>>
>>>> Just explain what you want to achieve.
>>>>
>>>> Best Regards
>>>> Sharad
>>>>
>>>>
>>>>
>>>> ----- Original Message -----
>>>> From: "Mitch Capper" <mitch.capper at gmail.com>
>>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, August 27, 2012 9:23 PM
>>>> Subject: Re: [Freeswitch-users] How to write an Application to
>>>> controlfreeSWITCH
>>>>
>>>>
>>>>  > look at the wiki for esl probably the easiest way if you need to you
>>>> > can also write fs modules in c++.
>>>> > ~mitch
>>>> >
>>>> > On Mon, Aug 27, 2012 at 8:19 AM, Subhash <gvvsubhashkumar at gmail.com>
>>>> > wrote:
>>>> >>
>>>> >> Hi All,
>>>> >>
>>>> >>        I am new to freeSWITCH platform.
>>>> >>
>>>> >>       I want to write an application using c++ to control freeSWITCH
>>>> so
>>>> >> please guide me.
>>>> >>
>>>> >>       Thanks in advance.
>>>> >>
>>>> >> Thanks,
>>>> >> Subhash.
>>>> >>
>>>> >>
>>>> _________________________________________________________________________
>>>> >> Professional FreeSWITCH Consulting Services:
>>>> >> consulting at freeswitch.org
>>>> >> http://www.freeswitchsolutions.com
>>>> >>
>>>> >> 
>>>> >> 
>>>> >>
>>>> >> Official FreeSWITCH Sites
>>>> >> http://www.freeswitch.org
>>>> >> http://wiki.freeswitch.org
>>>> >> http://www.cluecon.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
>>>> >>
>>>> >
>>>> >
>>>> >
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Aug 30 20:16:41 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 09:16:41 -0700
Subject: [Freeswitch-users] Prevent A leg from hangup after bridge with
 inbound ESL socket
In-Reply-To: <569384504C492C4580E88B5D54DFEAEA30CAFD3530@jjex01.jajah.dublin>
References: <569384504C492C4580E88B5D54DFEAEA30CAFD33B0@jjex01.jajah.dublin>
	<CAKzWOxW+jCoFfd-e0GkBDvsez4f66xqDLnzpYM6-QesJ+3Md7g@mail.gmail.com>
	<569384504C492C4580E88B5D54DFEAEA30CAFD3530@jjex01.jajah.dublin>
Message-ID: <CAKzWOxXWVgGyKRSG3ZiXhm4GGf9WbqSs3ABFWcmOWqdVOAy37g@mail.gmail.com>

If that's the case then you also need ignore_early_media=true:
<action application="bridge"
data="{ignore_early_media=true}sofia/internal/1001@$${domain}"/>

If you don't ignore early media then the bridge app assumes that when it
receives media from the far end that the bridge is "successful" even if you
don't actually get a 200OK. The caveat is that since you're ignoring early
media (i.e. ringing) from the B leg that you will need to supply some sort
of ringing indicator to the A leg. The good news is that you can do
whatever you want; just use the ring_back chan var.

-MC

On Thu, Aug 30, 2012 at 12:07 AM, Alex Massover <alex at jajah.com> wrote:

> Hi Michael,****
>
> ** **
>
> Thanks, that works in scenario when B legs response with 100 and then
> let's say 486. But if B legs do ringing, i.e. 100, 180/183, 486 it doesn't
> work.****
>
> ** **
>
> I found this in wiki "By the way, you'll be unable to rewrite the hangup
> cause for a bridge that gets a 180 or 183 packet from the gateway before
> getting a 4xx, 5xx or 6xx packet (because those bridges don't 'fail')."***
> *
>
> ** **
>
> I understand that continue_on_fail won't help with this scenario. I see
> that's a popular topic in the list, but nobody got a solution.****
>
> ** **
>
> BR, Alex.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Wednesday, August 29, 2012 6:21 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Prevent A leg from hangup after bridge
> with inbound ESL socket****
>
> ** **
>
> Try this:
> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>
> -MC****
>
> On Wed, Aug 29, 2012 at 6:57 AM, Alex Massover <alex at jajah.com> wrote:****
>
> Hi,****
>
>  ****
>
> I have a very simple dialplan that just do park for incoming calls. All
> rest of leg management is done via ESL inbound socket.****
>
>  ****
>
> I'm trying to do the same behavior like in this dialplan example, but from
> ESL inbound socket:****
>
> <action application="set" data="hangup_after_bridge=false"/> ****
>
>    <action application="bridge" data="sofia/internal/1001@$${domain}"/>***
> *
>
>    <action application="bridge" data="sofia/internal/1002@$${domain}"/>***
> *
>
>  ****
>
> The problem is with bridge API, if B leg doesn't answer (e.g. 404, or
> busy), A leg disconnects. But I'm trying to prevent A leg from
> disconnecting in order to do bridge to other place.****
>
>  ****
>
> Looks like hangup_after_bridge=false, park_after_bridge=true,
> transfer_after_bridge etc don't have any effect when bridge done from
> inbound socket. A leg disconnects always.****
>
>  ****
>
> Is there any way to keep A leg after bridge with inbound socket? I'm aware
> of originate, but prefer to user bridge.****
>
>  ****
>
>  ****
>
>  ****
>
>  ****
>
> --****
>
> Best Regards,****
>
> Alex Massover****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
> ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Aug 30 20:19:21 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 09:19:21 -0700
Subject: [Freeswitch-users] Best way to stream record session?
In-Reply-To: <000001cd86a8$75efcd10$61cf6730$@207me.com>
References: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FC202C@BLUPRD0711MB413.namprd07.prod.outlook.com>
	<000001cd86a8$75efcd10$61cf6730$@207me.com>
Message-ID: <CAKzWOxXDTUGWv4OitdRozX_DKWDmptiVwaBH8D+Bz2XDfN-umw@mail.gmail.com>

On Thu, Aug 30, 2012 at 5:10 AM, Stephen Dame <sdame at 207me.com> wrote:

> Mod_shout  configured  to an icecast server works well for this.   The wiki
> has examples.
>
+1 - this works very well in my experience.
-MC
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From mike at jerris.com  Thu Aug 30 20:19:39 2012
From: mike at jerris.com (Michael Jerris)
Date: Thu, 30 Aug 2012 12:19:39 -0400
Subject: [Freeswitch-users] Freeswitch - Stable Version
In-Reply-To: <CAKV-jFwetO7bc4OoRjovtmVFRDN0sq7UAQpi6czBYwvqxb3RoQ@mail.gmail.com>
References: <CAFTa0yd5YKw+_n_4DVEo73Hex9artpxCC-JreW1QL7BEtc9TZQ@mail.gmail.com>
	<CAPcNkde5qjzC4kg4R+3ZPKWzMwiFChXe_+u19mESgm1P+DoKbQ@mail.gmail.com>
	<12792.1346256601@ccs.covici.com>
	<CAPcNkddN_LZj8pnhWYEc+DPEBCT9NT3UyTRGV+3M_xiOjKJeBA@mail.gmail.com>
	<17159.1346258336@ccs.covici.com>
	<CAPcNkde9C9Eety1LD5SE-b91A=1Btudoum2eMjPen9rsrhh4aA@mail.gmail.com>
	<CAKV-jFwetO7bc4OoRjovtmVFRDN0sq7UAQpi6czBYwvqxb3RoQ@mail.gmail.com>
Message-ID: <D97C254E-43A0-4A92-98D8-F8CD3F9B21A9@jerris.com>

I don't have an issue with debian.  Just the people who use it.

On Aug 29, 2012, at 9:59 PM, Brian Foster <bdfoster at endigotech.com> wrote:

> Sorry I have to do this...
> 
> Go Debian!!!
> 
> Brian Foster
> Endigo Computer LLC
> 

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From msc at freeswitch.org  Thu Aug 30 20:24:28 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 09:24:28 -0700
Subject: [Freeswitch-users] how to play a prompt after connection
In-Reply-To: <k1nn6g$kd6$1@ger.gmane.org>
References: <23F564DC-911A-4D5B-A63F-D420C755B216@beachdognet.com>
	<F62AB8C5-4184-4937-9AE0-27EF32F000ED@beachdognet.com>
	<k1nn6g$kd6$1@ger.gmane.org>
Message-ID: <CAKzWOxW7HSWv5L28QenfCXbTsWYBSXkT+C0R+FNc5tMunuvdvw@mail.gmail.com>

I would try to get this working w/ just raw XML dialplan and take Lua out
of the equation. Once you get it working then migrate into a Lua script,
assuming you even need to use it. Personally, I try not to overuse Lua (or
any other dp scripting language) and keep as much in the XML dialplan as
possible.

Lab it up just with the dialplan using as simple a config as possible.
Capture logs and post to pastebin.freeswitch.org along with your relevant
configs (dialplan, conference) and some of the gang here will see if they
can help analyze what's going on.

-MC

On Thu, Aug 30, 2012 at 5:44 AM, Dave Horton <daveh at beachdognet.com> wrote:

> I did try that suggestion (api_on_answer uuidbroadcast) and it worked
> fine if I played it to the a leg, but when I tried playing to the b leg
> the prompt was incredibly choppy and slow, and delayed the cutthru of a
> and b legs by a long time.  Interestingly, the prompt doesn't even
> start playing to the b party until some RTP is received from b for some
> reason, and then it sort of slowly chuffs out the prompt in small
> choppy chunks.
>
> I then tried Michael's follow on suggestion of using
> bridge_pre_execute_bleg_app, but that didn't work at all -- no prompt
> heard.
>
> I then decided to try the outdial as a bridged conference, and then
> play a prompt to the conference after the b party picks up.  This may
> actually be the best option for me as I need the prompt to be saved in
> the recording of the call, and I am guessing when I stream it out on a
> bridged call it may not show up in the recording.
>
> However, I had no luck getting a bridged conference outdial to work at
> all -- the outdial does work successfully, but a and b parties don't
> hear each other so one must not be in the conference.  And then when
> the b party hangs up the a party is not automatically hung up, which
> should happen in a bridged conference if I understand things correctly.
>
> So, while I am still interested in what is causing the prompt on the
> bridge call to sound so bad when played to the b leg, perhaps what is
> more important to me now is getting the bridged conference to work.
> Here is what I am doing, in case anyone can point out the problem:
>
> session:execute("set","progress_timeout=60") ;
> session:execute("set","call_timeout=120") ;
> session:setVariable("hangup_after_bridge","true") ;
> session:execute("conference","bridge:_uuid_ at simple
> :sofia/normal_customer/${outdial}@${egress_gateway}")
> ;
> session:setVariable("conference_enter_sound",this_call_is_being_recorded) ;
>
> (I created a conference profile called 'simple' that turns off caller
> controls; interestingly, that doesn't seem to be picked up either as
> the debug logging shows it attaching the default caller controls)
>
> Dave
>
> On 2012-08-29 11:24:00 +0000, Dave Horton said:
>
> >>
> >> Set api_on_answer to execute
> >> uuid_broadcast<
> >> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast
> >>>
> >> which
> >> will let you pick which legs to play the recording to.
> >> You can pass in the UUID with ${uuid}
> >>
> >> -Avi
> >>
> > I tried this, and it worked.....sort of.  If I play the prompt the A
> > leg it is fine -- caller hears the prompt, then is connected to called
> > party.  If I play it to the B leg, however, the prompt does not seem to
> > start playing at all until the B party actually starts talking, and
> > then the prompt is played extremely choppy and slow.....it takes about
> > 10 seconds to play a 4 second prompt.  Meanwhile, the caller and called
> > party can't hear each other.  Any idea what is going on and how to fix
> > it?  I am doing it in lua as follows:
> >
> > session:execute("export","nolocal:jitterbuffer_msec=100") ;
> > session:setVariable("RECORD_STEREO","true") ;
> > session:setVariable("aleg_uuid",uuid) ;
> > session:execute("record_session", outfile) ;
> > session:execute("set","progress_timeout=60") ;
> > session:execute("set","call_timeout=120") ;
> > session:setVariable("hangup_after_bridge","true") ;
> > session:execute("bridge","sofia/normal_customer/${outdial}@
> ${egress_gateway}")
> > ;
> > --session:execute("bridge","{api_on_answer='uuid_broadcast " .. uuid ..
> > " " .. this_call_is_being_recorded .. " " ..
> > "bleg'}sofia/normal_customer/${outdial}@${egress_gateway}") ;
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Aug 30 20:29:57 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 09:29:57 -0700
Subject: [Freeswitch-users] Eavesdrop with bind_digit_action
In-Reply-To: <19C8F06C-3FCD-4FBA-84E4-B656E7846479@kickasspixels.com>
References: <mailman.17959.1346332495.13443.freeswitch-users@lists.freeswitch.org>
	<19C8F06C-3FCD-4FBA-84E4-B656E7846479@kickasspixels.com>
Message-ID: <CAKzWOxVBbHKP4QPGDy7fMeX6F2ahy6ZbVDKkARVzbD68dspDRQ@mail.gmail.com>

Malfunction! Need input! Can you throw some logs and config info on
pastebin? I think some folks might be interested in seeing this problem
solved and if you've already started the work I bet others would be willing
to assist.

-MC

On Thu, Aug 30, 2012 at 8:59 AM, Lon Baker <lon at kickasspixels.com> wrote:

> I'm trying to add the ability to screen voicemail being left, and allow
> the callee to pull the call back from voicemail with a DTMF. I thought I
> could use eavesdrop and bind_digit_action, but its not working.
>
> The bound DTMF is logged in the console, but the exec action is never
> triggered.
>
> Caller is correctly sent to voicemail
> Callee, who elected to eavesdrop, can hear the voicemail being left.
> Pressing 9 for the bind_digit_action, show up in the console.
>
> But, exec extension does not go off.
>
> Any help would be appreciated.
>
> --
> Lon
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From gvvsubhashkumar at gmail.com  Thu Aug 30 20:59:42 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Thu, 30 Aug 2012 22:29:42 +0530
Subject: [Freeswitch-users] DialPlan Generation
Message-ID: <CAFvodxXx0nN8kfg5V-E7M3oTVC34Sr_RJ-EgxzULQELLyZ+6tA@mail.gmail.com>

Hi,

     Which is the best way to create(static or dynamic) dialplan?

    Please suggest me.


Thanks,
Subhash.
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From vipkilla at gmail.com  Thu Aug 30 21:49:25 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Thu, 30 Aug 2012 13:49:25 -0400
Subject: [Freeswitch-users] DialPlan Generation
In-Reply-To: <CAFvodxXx0nN8kfg5V-E7M3oTVC34Sr_RJ-EgxzULQELLyZ+6tA@mail.gmail.com>
References: <CAFvodxXx0nN8kfg5V-E7M3oTVC34Sr_RJ-EgxzULQELLyZ+6tA@mail.gmail.com>
Message-ID: <CAC-LwPNiADX8be2M08e4otqFjzkwA22-KkDQvMYgWRoxKhwnwg@mail.gmail.com>

For dynamic dialplans, IMO outbound ESL is the *BEST* method but also
most advanced.
Other commonly used options methods are XML_CURL and LUA.

On Thu, Aug 30, 2012 at 12:59 PM, Subhash <gvvsubhashkumar at gmail.com> wrote:
> Hi,
>
>      Which is the best way to create(static or dynamic) dialplan?
>
>     Please suggest me.
>
>
> Thanks,
> Subhash.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 daveh at beachdognet.com  Thu Aug 30 22:59:34 2012
From: daveh at beachdognet.com (Dave Horton)
Date: Thu, 30 Aug 2012 14:59:34 -0400
Subject: [Freeswitch-users] how to play a prompt after connection
References: <23F564DC-911A-4D5B-A63F-D420C755B216@beachdognet.com>
	<F62AB8C5-4184-4937-9AE0-27EF32F000ED@beachdognet.com>
	<k1nn6g$kd6$1@ger.gmane.org>
	<CAKzWOxW7HSWv5L28QenfCXbTsWYBSXkT+C0R+FNc5tMunuvdvw@mail.gmail.com>
Message-ID: <k1od6m$ik$1@ger.gmane.org>

On 2012-08-30 16:24:28 +0000, Michael Collins said:

> Lab it up just with the dialplan using as simple a config as possible. 
> Capture logs and post to pastebin.freeswitch.org along with your 
> relevant configs (dialplan, conference) and some of the gang here will 
> see if they can help analyze what's going on.

Ok, I reproduced the scenario where it is a bridged call (not a 
conference) where I play the prompt after answer and get the extremely 
choppy audio (10 seconds or more to play a 4 second prompt).  The 
pastebin of the debug logging is at 
http://pastebin.freeswitch.org/19785, and the configuration I used is 
below.  If anyone can point out what might be causing this it would be 
appreciated.  Next I will try the conferencing version and paste those 
results

    <extension name="6174973896" continue="false">
      <condition field="source" expression="mod_sofia"/>
      <condition field="destination_number" expression="6174973896">
                <action application="set" data="call_limit=1"/>
                <action application="set" data="callFactor=1"/>
                <action application="set" data="customer=cps_play"/>
                <action application="ring_ready"/>
                <action application="pre_answer"/>
                <action application="start_dtmf"/>
                <action application="set" 
data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/>
                <action application="set" 
data="outfile=$${base_dir}/recordings/${call_id}.wav"/>
                <action application="export" 
data="nolocal:jitterbuffer_msec=100"/>
                <action application="set" data="RECORD_STEREO=true"/>
                <action application="set" data="aleg_uuid=${uuid}"/>
                <action application="set" 
data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/>
                <action application="set" 
data="outfile=$${base_dir}/recordings/${call_id}.wav"/>
                <action application="record_session" data="${outfile}"/>
                <action application="set" data="progress_timeout=60"/>
                <action application="set" data="call_timeout=120"/>
                <action application="set" 
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE"/>
                <action application="set" data="hangup_after_bridge=true"/>
                <action application="export" 
data="nolocal:api_on_answer=uuid_broadcast ${uuid} 
misc/this-call-is-being-recorded.wav bleg"/>
                <action application="bridge" 
data="sofia/normal_customer/15083084809@${egress_gateway}"/>

        </condition>
    </extension>





From freeswitch-users at vocalspace.com  Thu Aug 30 22:20:06 2012
From: freeswitch-users at vocalspace.com (Phillip Boles)
Date: Thu, 30 Aug 2012 13:20:06 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging in
	CS_SOFT_EXEC state
Message-ID: <FB1F0348-B48D-44FA-821D-1473E4EEDCE1@vocalspace.com>

	
	var session =  ManagedSession.OriginateHandleHangup(s, dialString, ts, (y) => {
		try {
			Log.WriteLine(LogLevel.Info, "Hanging UP: "+ y.GetUuid());
			y.SetAutoHangup(true);
			y.destroy();

		} catch( Exception ) {
			Log.WriteLine(LogLevel.Critical, "Exception While Trying to handup");	
		}
	});
			
	
2012-08-28 10:49:12.156172 [INFO] switch_cpp.cpp:1227 Dialing
2012-08-28 10:49:12.156172 [DEBUG] switch_ivr_originate.c:1947 Parsing global variables
2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [ignore_early_media]=[true]
2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [origination_caller_id_number]=[2223334444]
2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [origination_caller_id_name]=[Vocalspace]
2012-08-28 10:49:12.156172 [NOTICE] switch_channel.c:926 New Channel sofia/external/XXXXXXXXXXX [e315f2e8-1fa8-4fd9-849b-f687dad8aed5]
2012-08-28 10:49:12.156172 [DEBUG] mod_sofia.c:4709 (sofia/external/XXXXXXXXXXX) State Change CS_NEW -> CS_INIT
2012-08-28 10:49:12.156172 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:12.156172 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_INIT
2012-08-28 10:49:12.156172 [DEBUG] switch_core_state_machine.c:421 (sofia/external/XXXXXXXXXXX) State INIT
2012-08-28 10:49:12.156172 [DEBUG] mod_sofia.c:85 sofia/external/XXXXXXXXXXX SOFIA INIT
2012-08-28 10:49:12.216161 [DEBUG] switch_nat.c:511 mapped public port 19628 protocol UDP to localport 19628
2012-08-28 10:49:12.256159 [DEBUG] switch_nat.c:511 mapped public port 19629 protocol UDP to localport 19629
2012-08-28 10:49:12.256159 [DEBUG] mod_sofia.c:129 (sofia/external/XXXXXXXXXXX) State Change CS_INIT -> CS_ROUTING
2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:421 (sofia/external/XXXXXXXXXXX) State INIT going to sleep
2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_ROUTING
2012-08-28 10:49:12.256159 [DEBUG] switch_channel.c:1887 (sofia/external/XXXXXXXXXXX) Callstate Change DOWN -> RINGING
2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:430 (sofia/external/XXXXXXXXXXX) State ROUTING
2012-08-28 10:49:12.256159 [DEBUG] mod_sofia.c:152 sofia/external/XXXXXXXXXXX SOFIA ROUTING
2012-08-28 10:49:12.256159 [DEBUG] switch_ivr_originate.c:67 (sofia/external/XXXXXXXXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:430 (sofia/external/XXXXXXXXXXX) State ROUTING going to sleep
2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_CONSUME_MEDIA
2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:449 (sofia/external/XXXXXXXXXXX) State CONSUME_MEDIA
2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:449 (sofia/external/XXXXXXXXXXX) State CONSUME_MEDIA going to sleep
2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:12.256159 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [calling][0]
2012-08-28 10:49:12.636131 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:12.636131 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:12.636131 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call" 
2012-08-28 10:49:12.636131 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [proceeding][180]
2012-08-28 10:49:12.636131 [NOTICE] sofia.c:5769 Ring-Ready sofia/external/XXXXXXXXXXX!
2012-08-28 10:49:14.655990 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:14.655990 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:14.655990 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call" 
2012-08-28 10:49:14.655990 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [proceeding][183]
2012-08-28 10:49:14.655990 [DEBUG] sofia.c:5688 Remote SDP:
v=0
o=FreeSWITCH 1346150407 1346150408 IN IP4 8.19.97.6
s=FreeSWITCH
c=IN IP4 8.19.97.6
t=0 0
m=audio 18534 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:4911 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:2995 Set Codec sofia/external/XXXXXXXXXXX PCMU/8000 20 ms 160 samples 64000 bits
2012-08-28 10:49:14.655990 [DEBUG] switch_core_codec.c:111 sofia/external/XXXXXXXXXXX Original read codec set to PCMU:0
2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:5025 Set 2833 dtmf send payload to 101
2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3244 AUDIO RTP [sofia/external/XXXXXXXXXXX] 10.10.80.33 port 19628 -> 8.19.97.6 port 18534 codec: 0 ms: 20
2012-08-28 10:49:14.655990 [DEBUG] switch_rtp.c:1676 Starting timer [soft] 160 bytes per 20ms
2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3508 Set 2833 dtmf send payload to 101
2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3514 Set 2833 dtmf receive payload to 101
2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3541 sofia/external/XXXXXXXXXXX Set rtp dtmf delay to 40
2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3547 Set comfort noise payload to 13
2012-08-28 10:49:14.655990 [NOTICE] sofia_glue.c:4052 Pre-Answer sofia/external/XXXXXXXXXXX!
2012-08-28 10:49:14.655990 [DEBUG] switch_channel.c:2986 (sofia/external/XXXXXXXXXXX) Callstate Change RINGING -> EARLY
2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:22.136467 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call" 
2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [completing][200]
2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5685 Duplicate SDP
v=0
o=FreeSWITCH 1346150407 1346150408 IN IP4 8.19.97.6
s=FreeSWITCH
c=IN IP4 8.19.97.6
t=0 0
m=audio 18534 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [ready][200]
2012-08-28 10:49:22.136467 [DEBUG] switch_channel.c:3245 (sofia/external/XXXXXXXXXXX) Callstate Change EARLY -> ACTIVE
2012-08-28 10:49:22.136467 [NOTICE] sofia.c:6352 Channel [sofia/external/XXXXXXXXXXX] has been answered
2012-08-28 10:49:22.136467 [DEBUG] switch_ivr_originate.c:3330 Originate Resulted in Success: [sofia/external/XXXXXXXXXXX]
2012-08-28 10:49:22.136467 [DEBUG] switch_cpp.cpp:1083 (sofia/external/XXXXXXXXXXX) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE
2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_SOFT_EXECUTE
2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:443 (sofia/external/XXXXXXXXXXX) State SOFT_EXECUTE
2012-08-28 10:49:22.136467 [DEBUG] mod_sofia.c:592 SOFIA SOFT_EXECUTE
2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:261 sofia/external/XXXXXXXXXXX Standard SOFT_EXECUTE
2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:443 (sofia/external/XXXXXXXXXXX) State SOFT_EXECUTE going to sleep
2012-08-28 10:49:22.136467 [DEBUG] switch_cpp.cpp:1172 CoreSession::seHangupHook, hangup_func: (nil)
2012-08-28 10:49:22.136467 [INFO] switch_cpp.cpp:1227 Connected
2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Session Ready
2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Call Initialized
2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Trying To Run
2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Running
2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Checking for file /svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw
2012-08-28 10:49:23.160407 [ERR] mod_sndfile.c:197 Error Opening File [/svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw] [System error : No such file or directory.]
2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Transcoding xfile /svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw
2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Sending MQ Message
2012-08-28 10:49:23.296392 [NOTICE] switch_cpp.cpp:1227 Creating a channel...
2012-08-28 10:49:23.296392 [NOTICE] switch_cpp.cpp:1227 Defining Exchange FS...
2012-08-28 10:49:23.316389 [NOTICE] switch_cpp.cpp:1227 Binding Exchange and Queue..
2012-08-28 10:49:23.336389 [NOTICE] switch_cpp.cpp:1227 Sending message..
2012-08-28 10:49:23.356388 [NOTICE] switch_cpp.cpp:1227 shout://www.product.com/files/media/e53bc5bf-4e00-ab8d-4e4d-b201699628d1.mp3
2012-08-28 10:49:25.916204 [DEBUG] mod_shout.c:472 Read Thread Done
2012-08-28 10:49:26.356173 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
2012-08-28 10:49:26.596156 [DEBUG] switch_rtp.c:3252 Correct ip/port confirmed.
2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:27.296108 [DEBUG] switch_channel.c:2849 (sofia/external/XXXXXXXXXXX) Callstate Change ACTIVE -> HANGUP
2012-08-28 10:49:27.296108 [NOTICE] sofia.c:694 Hangup sofia/external/XXXXXXXXXXX [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
2012-08-28 10:49:27.296108 [DEBUG] switch_channel.c:2872 Send signal sofia/external/XXXXXXXXXXX [KILL]
2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1227 AppFunction is in hangupCallback.
2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_HANGUP
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:622 (sofia/external/XXXXXXXXXXX) State HANGUP
2012-08-28 10:49:27.296108 [DEBUG] mod_sofia.c:473 Channel sofia/external/XXXXXXXXXXX hanging up, cause: NORMAL_CLEARING
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:47 sofia/external/XXXXXXXXXXX Standard HANGUP, cause: NORMAL_CLEARING
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:622 (sofia/external/XXXXXXXXXXX) State HANGUP going to sleep
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:413 (sofia/external/XXXXXXXXXXX) State Change CS_HANGUP -> CS_REPORTING
2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1227 AppFunction is in hangupCallback.
2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_REPORTING
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682 (sofia/external/XXXXXXXXXXX) State REPORTING
2012-08-28 10:49:27.296108 [INFO] switch_cpp.cpp:1227 Handing UP: e315f2e8-1fa8-4fd9-849b-f687dad8aed5
2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1007 sofia/external/XXXXXXXXXXX destroy/unlink session from object
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:79 sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682 (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:407 (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY
2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1424 Session 4 (sofia/external/XXXXXXXXXXX) Locked, Waiting on external entities
2012-08-28 10:49:27.316110 [DEBUG] switch_ivr_play_say.c:1682 done playing file shout://www.product.com/files/media/e53bc5bf-4e00-ab8d-4e4d-b201699628d1.mp3

freeswitch at fs03.int.colo> show channels

uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
e315f2e8-1fa8-4fd9-849b-f687dad8aed5,outbound,2012-08-28 10:49:12,1346168952,sofia/external/XXXXXXXXXXX,CS_SOFT_EXECUTE,Vocalspace,2223334444,,XXXXXXXXXXX,,,,default,PCMU,8000,64000,PCMU,8000,64000,,fs03.int.colo,,,ACTIVE,Outbound Call,XXXXXXXXXXX,,,,

1 total.

2012-08-28 10:58:10.256585 [DEBUG] switch_nat.c:511 mapped public port 5060 protocol TCP to localport 5060
2012-08-28 10:58:10.296583 [DEBUG] switch_nat.c:511 mapped public port 5080 protocol UDP to localport 5080
2012-08-28 10:58:10.336580 [DEBUG] switch_nat.c:511 mapped public port 5080 protocol TCP to localport 5080
2012-08-28 10:58:10.396576 [DEBUG] switch_nat.c:511 mapped public port 19628 protocol UDP to localport 19628
2012-08-28 10:58:10.436573 [DEBUG] switch_nat.c:511 mapped public port 19629 protocol UDP to localport 19629
2012-08-28 10:58:10.476570 [DEBUG] switch_nat.c:511 mapped public port 22013 protocol UDP to localport 22013
2012-08-28 10:58:10.536566 [DEBUG] switch_nat.c:511 mapped public port 22191 protocol UDP to localport 22191
2012-08-28 10:58:10.576563 [DEBUG] switch_nat.c:511 mapped public port 25326 protocol UDP to localport 25326
2012-08-28 10:58:10.636559 [DEBUG] switch_nat.c:511 mapped public port 25327 protocol UDP to localport 25327
Loading /usr/lib64/freeswitch/mod/managed/Enyim.Caching.dll from domain Enyim.Caching.dll_63
2012-08-28 11:00:02.097780 [DEBUG] switch_cpp.cpp:1227 Assembly Enyim.Caching, Version=1.2.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/Jint.dll from domain Jint.dll_64
2012-08-28 11:00:02.316760 [DEBUG] switch_cpp.cpp:1227 Assembly Jint, Version=0.9.2.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/log4net.dll from domain log4net.dll_65
2012-08-28 11:00:02.536745 [DEBUG] switch_cpp.cpp:1227 Assembly log4net, Version=1.2.10.0, Culture=neutral, PublicKeyToken=1b44e1d426115821 doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/Antlr3.Runtime.dll from domain Antlr3.Runtime.dll_66
2012-08-28 11:00:02.756720 [DEBUG] switch_cpp.cpp:1227 Assembly Antlr3.Runtime, Version=3.3.1.7705, Culture=neutral, PublicKeyToken=eb42632606e9261f doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/RabbitMQ.Client.dll from domain RabbitMQ.Client.dll_67
2012-08-28 11:00:02.996708 [DEBUG] switch_cpp.cpp:1227 Assembly RabbitMQ.Client, Version=0.0.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/ManagedEsl.dll from domain ManagedEsl.dll_68
2012-08-28 11:00:03.216689 [DEBUG] switch_cpp.cpp:1227 Assembly ManagedEsl, Version=1.0.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/FreeSWITCH.Managed.dll from domain FreeSWITCH.Managed.dll_69
2012-08-28 11:00:03.436678 [DEBUG] switch_cpp.cpp:1227 Assembly FreeSWITCH.Managed, Version=1.0.5.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/FreeSwitch.EventSocket.dll from domain FreeSwitch.EventSocket.dll_70
2012-08-28 11:00:03.657671 [DEBUG] switch_cpp.cpp:1227 Assembly FreeSwitch.EventSocket, Version=1.0.0.0, Culture=neutral, PublicKeyToken=94d8a22cdb4b7713 doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/Newtonsoft.Json.dll from domain Newtonsoft.Json.dll_71
2012-08-28 11:00:03.876655 [DEBUG] switch_cpp.cpp:1227 Assembly Newtonsoft.Json, Version=4.5.0.0, Culture=neutral, PublicKeyToken=30ad4fe6b2a6aeed doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/Freeswitch.UUID.dll from domain Freeswitch.UUID.dll_72
2012-08-28 11:00:04.096628 [DEBUG] switch_cpp.cpp:1227 Assembly Freeswitch.UUID, Version=1.0.0.0, Culture=neutral, PublicKeyToken=f758751257b2cdaf doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/Product.dll from domain Product.dll_73

freeswitch at fs03.int.colo> show channels


0 total.

	
	var session =  ManagedSession.OriginateHandleHangup(s, dialString, ts, (y) => {
		try {
			Log.WriteLine(LogLevel.Info, "Hanging UP: "+ y.GetUuid());
			y.SetAutoHangup(true);
			y.destroy();

		} catch( Exception ) {
			Log.WriteLine(LogLevel.Critical, "Exception While Trying to handup");	
		}
	});
			
	
2012-08-28 10:49:12.156172 [INFO] switch_cpp.cpp:1227 Dialing
2012-08-28 10:49:12.156172 [DEBUG] switch_ivr_originate.c:1947 Parsing global variables
2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [ignore_early_media]=[true]
2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [origination_caller_id_number]=[2223334444]
2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [origination_caller_id_name]=[Vocalspace]
2012-08-28 10:49:12.156172 [NOTICE] switch_channel.c:926 New Channel sofia/external/XXXXXXXXXXX [e315f2e8-1fa8-4fd9-849b-f687dad8aed5]
2012-08-28 10:49:12.156172 [DEBUG] mod_sofia.c:4709 (sofia/external/XXXXXXXXXXX) State Change CS_NEW -> CS_INIT
2012-08-28 10:49:12.156172 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:12.156172 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_INIT
2012-08-28 10:49:12.156172 [DEBUG] switch_core_state_machine.c:421 (sofia/external/XXXXXXXXXXX) State INIT
2012-08-28 10:49:12.156172 [DEBUG] mod_sofia.c:85 sofia/external/XXXXXXXXXXX SOFIA INIT
2012-08-28 10:49:12.216161 [DEBUG] switch_nat.c:511 mapped public port 19628 protocol UDP to localport 19628
2012-08-28 10:49:12.256159 [DEBUG] switch_nat.c:511 mapped public port 19629 protocol UDP to localport 19629
2012-08-28 10:49:12.256159 [DEBUG] mod_sofia.c:129 (sofia/external/XXXXXXXXXXX) State Change CS_INIT -> CS_ROUTING
2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:421 (sofia/external/XXXXXXXXXXX) State INIT going to sleep
2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_ROUTING
2012-08-28 10:49:12.256159 [DEBUG] switch_channel.c:1887 (sofia/external/XXXXXXXXXXX) Callstate Change DOWN -> RINGING
2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:430 (sofia/external/XXXXXXXXXXX) State ROUTING
2012-08-28 10:49:12.256159 [DEBUG] mod_sofia.c:152 sofia/external/XXXXXXXXXXX SOFIA ROUTING
2012-08-28 10:49:12.256159 [DEBUG] switch_ivr_originate.c:67 (sofia/external/XXXXXXXXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:430 (sofia/external/XXXXXXXXXXX) State ROUTING going to sleep
2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_CONSUME_MEDIA
2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:449 (sofia/external/XXXXXXXXXXX) State CONSUME_MEDIA
2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:449 (sofia/external/XXXXXXXXXXX) State CONSUME_MEDIA going to sleep
2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:12.256159 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [calling][0]
2012-08-28 10:49:12.636131 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:12.636131 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:12.636131 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call" 
2012-08-28 10:49:12.636131 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [proceeding][180]
2012-08-28 10:49:12.636131 [NOTICE] sofia.c:5769 Ring-Ready sofia/external/XXXXXXXXXXX!
2012-08-28 10:49:14.655990 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:14.655990 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:14.655990 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call" 
2012-08-28 10:49:14.655990 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [proceeding][183]
2012-08-28 10:49:14.655990 [DEBUG] sofia.c:5688 Remote SDP:
v=0
o=FreeSWITCH 1346150407 1346150408 IN IP4 8.19.97.6
s=FreeSWITCH
c=IN IP4 8.19.97.6
t=0 0
m=audio 18534 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:4911 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:2995 Set Codec sofia/external/XXXXXXXXXXX PCMU/8000 20 ms 160 samples 64000 bits
2012-08-28 10:49:14.655990 [DEBUG] switch_core_codec.c:111 sofia/external/XXXXXXXXXXX Original read codec set to PCMU:0
2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:5025 Set 2833 dtmf send payload to 101
2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3244 AUDIO RTP [sofia/external/XXXXXXXXXXX] 10.10.80.33 port 19628 -> 8.19.97.6 port 18534 codec: 0 ms: 20
2012-08-28 10:49:14.655990 [DEBUG] switch_rtp.c:1676 Starting timer [soft] 160 bytes per 20ms
2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3508 Set 2833 dtmf send payload to 101
2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3514 Set 2833 dtmf receive payload to 101
2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3541 sofia/external/XXXXXXXXXXX Set rtp dtmf delay to 40
2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3547 Set comfort noise payload to 13
2012-08-28 10:49:14.655990 [NOTICE] sofia_glue.c:4052 Pre-Answer sofia/external/XXXXXXXXXXX!
2012-08-28 10:49:14.655990 [DEBUG] switch_channel.c:2986 (sofia/external/XXXXXXXXXXX) Callstate Change RINGING -> EARLY
2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:22.136467 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call" 
2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [completing][200]
2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5685 Duplicate SDP
v=0
o=FreeSWITCH 1346150407 1346150408 IN IP4 8.19.97.6
s=FreeSWITCH
c=IN IP4 8.19.97.6
t=0 0
m=audio 18534 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [ready][200]
2012-08-28 10:49:22.136467 [DEBUG] switch_channel.c:3245 (sofia/external/XXXXXXXXXXX) Callstate Change EARLY -> ACTIVE
2012-08-28 10:49:22.136467 [NOTICE] sofia.c:6352 Channel [sofia/external/XXXXXXXXXXX] has been answered
2012-08-28 10:49:22.136467 [DEBUG] switch_ivr_originate.c:3330 Originate Resulted in Success: [sofia/external/XXXXXXXXXXX]
2012-08-28 10:49:22.136467 [DEBUG] switch_cpp.cpp:1083 (sofia/external/XXXXXXXXXXX) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE
2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_SOFT_EXECUTE
2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:443 (sofia/external/XXXXXXXXXXX) State SOFT_EXECUTE
2012-08-28 10:49:22.136467 [DEBUG] mod_sofia.c:592 SOFIA SOFT_EXECUTE
2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:261 sofia/external/XXXXXXXXXXX Standard SOFT_EXECUTE
2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:443 (sofia/external/XXXXXXXXXXX) State SOFT_EXECUTE going to sleep
2012-08-28 10:49:22.136467 [DEBUG] switch_cpp.cpp:1172 CoreSession::seHangupHook, hangup_func: (nil)
2012-08-28 10:49:22.136467 [INFO] switch_cpp.cpp:1227 Connected
2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Session Ready
2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Call Initialized
2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Trying To Run
2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Running
2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Checking for file /svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw
2012-08-28 10:49:23.160407 [ERR] mod_sndfile.c:197 Error Opening File [/svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw] [System error : No such file or directory.]
2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Transcoding xfile /svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw
2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Sending MQ Message
2012-08-28 10:49:23.296392 [NOTICE] switch_cpp.cpp:1227 Creating a channel...
2012-08-28 10:49:23.296392 [NOTICE] switch_cpp.cpp:1227 Defining Exchange FS...
2012-08-28 10:49:23.316389 [NOTICE] switch_cpp.cpp:1227 Binding Exchange and Queue..
2012-08-28 10:49:23.336389 [NOTICE] switch_cpp.cpp:1227 Sending message..
2012-08-28 10:49:23.356388 [NOTICE] switch_cpp.cpp:1227 shout://www.product.com/files/media/e53bc5bf-4e00-ab8d-4e4d-b201699628d1.mp3
2012-08-28 10:49:25.916204 [DEBUG] mod_shout.c:472 Read Thread Done
2012-08-28 10:49:26.356173 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
2012-08-28 10:49:26.596156 [DEBUG] switch_rtp.c:3252 Correct ip/port confirmed.
2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:27.296108 [DEBUG] switch_channel.c:2849 (sofia/external/XXXXXXXXXXX) Callstate Change ACTIVE -> HANGUP
2012-08-28 10:49:27.296108 [NOTICE] sofia.c:694 Hangup sofia/external/XXXXXXXXXXX [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
2012-08-28 10:49:27.296108 [DEBUG] switch_channel.c:2872 Send signal sofia/external/XXXXXXXXXXX [KILL]
2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1227 AppFunction is in hangupCallback.
2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_HANGUP
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:622 (sofia/external/XXXXXXXXXXX) State HANGUP
2012-08-28 10:49:27.296108 [DEBUG] mod_sofia.c:473 Channel sofia/external/XXXXXXXXXXX hanging up, cause: NORMAL_CLEARING
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:47 sofia/external/XXXXXXXXXXX Standard HANGUP, cause: NORMAL_CLEARING
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:622 (sofia/external/XXXXXXXXXXX) State HANGUP going to sleep
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:413 (sofia/external/XXXXXXXXXXX) State Change CS_HANGUP -> CS_REPORTING
2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1227 AppFunction is in hangupCallback.
2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_REPORTING
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682 (sofia/external/XXXXXXXXXXX) State REPORTING
2012-08-28 10:49:27.296108 [INFO] switch_cpp.cpp:1227 Handing UP: e315f2e8-1fa8-4fd9-849b-f687dad8aed5
2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1007 sofia/external/XXXXXXXXXXX destroy/unlink session from object
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:79 sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682 (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:407 (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY
2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1424 Session 4 (sofia/external/XXXXXXXXXXX) Locked, Waiting on external entities
2012-08-28 10:49:27.316110 [DEBUG] switch_ivr_play_say.c:1682 done playing file shout://www.product.com/files/media/e53bc5bf-4e00-ab8d-4e4d-b201699628d1.mp3

freeswitch at fs03.int.colo> show channels

uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
e315f2e8-1fa8-4fd9-849b-f687dad8aed5,outbound,2012-08-28 10:49:12,1346168952,sofia/external/XXXXXXXXXXX,CS_SOFT_EXECUTE,Vocalspace,2223334444,,XXXXXXXXXXX,,,,default,PCMU,8000,64000,PCMU,8000,64000,,fs03.int.colo,,,ACTIVE,Outbound Call,XXXXXXXXXXX,,,,

1 total.

2012-08-28 10:58:10.256585 [DEBUG] switch_nat.c:511 mapped public port 5060 protocol TCP to localport 5060
2012-08-28 10:58:10.296583 [DEBUG] switch_nat.c:511 mapped public port 5080 protocol UDP to localport 5080
2012-08-28 10:58:10.336580 [DEBUG] switch_nat.c:511 mapped public port 5080 protocol TCP to localport 5080
2012-08-28 10:58:10.396576 [DEBUG] switch_nat.c:511 mapped public port 19628 protocol UDP to localport 19628
2012-08-28 10:58:10.436573 [DEBUG] switch_nat.c:511 mapped public port 19629 protocol UDP to localport 19629
2012-08-28 10:58:10.476570 [DEBUG] switch_nat.c:511 mapped public port 22013 protocol UDP to localport 22013
2012-08-28 10:58:10.536566 [DEBUG] switch_nat.c:511 mapped public port 22191 protocol UDP to localport 22191
2012-08-28 10:58:10.576563 [DEBUG] switch_nat.c:511 mapped public port 25326 protocol UDP to localport 25326
2012-08-28 10:58:10.636559 [DEBUG] switch_nat.c:511 mapped public port 25327 protocol UDP to localport 25327
Loading /usr/lib64/freeswitch/mod/managed/Enyim.Caching.dll from domain Enyim.Caching.dll_63
2012-08-28 11:00:02.097780 [DEBUG] switch_cpp.cpp:1227 Assembly Enyim.Caching, Version=1.2.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/Jint.dll from domain Jint.dll_64
2012-08-28 11:00:02.316760 [DEBUG] switch_cpp.cpp:1227 Assembly Jint, Version=0.9.2.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/log4net.dll from domain log4net.dll_65
2012-08-28 11:00:02.536745 [DEBUG] switch_cpp.cpp:1227 Assembly log4net, Version=1.2.10.0, Culture=neutral, PublicKeyToken=1b44e1d426115821 doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/Antlr3.Runtime.dll from domain Antlr3.Runtime.dll_66
2012-08-28 11:00:02.756720 [DEBUG] switch_cpp.cpp:1227 Assembly Antlr3.Runtime, Version=3.3.1.7705, Culture=neutral, PublicKeyToken=eb42632606e9261f doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/RabbitMQ.Client.dll from domain RabbitMQ.Client.dll_67
2012-08-28 11:00:02.996708 [DEBUG] switch_cpp.cpp:1227 Assembly RabbitMQ.Client, Version=0.0.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/ManagedEsl.dll from domain ManagedEsl.dll_68
2012-08-28 11:00:03.216689 [DEBUG] switch_cpp.cpp:1227 Assembly ManagedEsl, Version=1.0.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/FreeSWITCH.Managed.dll from domain FreeSWITCH.Managed.dll_69
2012-08-28 11:00:03.436678 [DEBUG] switch_cpp.cpp:1227 Assembly FreeSWITCH.Managed, Version=1.0.5.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/FreeSwitch.EventSocket.dll from domain FreeSwitch.EventSocket.dll_70
2012-08-28 11:00:03.657671 [DEBUG] switch_cpp.cpp:1227 Assembly FreeSwitch.EventSocket, Version=1.0.0.0, Culture=neutral, PublicKeyToken=94d8a22cdb4b7713 doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/Newtonsoft.Json.dll from domain Newtonsoft.Json.dll_71
2012-08-28 11:00:03.876655 [DEBUG] switch_cpp.cpp:1227 Assembly Newtonsoft.Json, Version=4.5.0.0, Culture=neutral, PublicKeyToken=30ad4fe6b2a6aeed doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/Freeswitch.UUID.dll from domain Freeswitch.UUID.dll_72
2012-08-28 11:00:04.096628 [DEBUG] switch_cpp.cpp:1227 Assembly Freeswitch.UUID, Version=1.0.0.0, Culture=neutral, PublicKeyToken=f758751257b2cdaf doesn't reference FreeSWITCH.Managed, not loading.
Loading /usr/lib64/freeswitch/mod/managed/Product.dll from domain Product.dll_73

freeswitch at fs03.int.colo> show channels


0 total.

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From daveh at beachdognet.com  Thu Aug 30 23:31:46 2012
From: daveh at beachdognet.com (Dave Horton)
Date: Thu, 30 Aug 2012 15:31:46 -0400
Subject: [Freeswitch-users] how to play a prompt after connection
References: <23F564DC-911A-4D5B-A63F-D420C755B216@beachdognet.com>
	<F62AB8C5-4184-4937-9AE0-27EF32F000ED@beachdognet.com>
	<k1nn6g$kd6$1@ger.gmane.org>
	<CAKzWOxW7HSWv5L28QenfCXbTsWYBSXkT+C0R+FNc5tMunuvdvw@mail.gmail.com>
	<k1od6m$ik$1@ger.gmane.org>
Message-ID: <k1of31$hhj$1@ger.gmane.org>

On 2012-08-30 18:59:34 +0000, Dave Horton said:

> Next I will try the conferencing version and paste those
> results

Below is the configuration I used with the bridged conferencing. 
Something must be wrong because there is no audio path between the a 
and b legs; also, when b hangs up a stays connected.

Pastebin is at http://pastebin.freeswitch.org/19786 and configuration 
is below.  I'd welcome any thoughts on what is wrong...


    <extension name="6174973896" continue="false">
      <condition field="source" expression="mod_sofia"/>
      <condition field="destination_number" expression="6174973896">
                <action application="set" data="call_limit=1"/>
                <action application="set" data="callFactor=1"/>
                <action application="set" data="customer=cps_play"/>
                <action application="ring_ready"/>
                <action application="pre_answer"/>
                <action application="start_dtmf"/>
                <action application="set" 
data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/>
                <action application="set" 
data="outfile=$${base_dir}/recordings/${call_id}.wav"/>
                <action application="export" 
data="nolocal:jitterbuffer_msec=100"/>
                <action application="set" data="RECORD_STEREO=true"/>
                <action application="set" data="aleg_uuid=${uuid}"/>
                <action application="set" 
data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/>
                <action application="set" 
data="outfile=$${base_dir}/recordings/${call_id}.wav"/>
                <action application="record_session" data="${outfile}"/>
                <action application="set" data="progress_timeout=60"/>
                <action application="set" data="call_timeout=120"/>
                <action application="set" 
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE"/>
                <action application="set" data="hangup_after_bridge=true"/>
                <action application="conference" 
data="bridge:_uuid_ at simple:sofia/normal_customer/15083084809@${egress_gateway}"/> 

                <action application="set" 
data="conference_enter_sound=misc/this-call-is-being-recorded"/>

        </condition>
    </extension>




From msc at freeswitch.org  Thu Aug 30 23:37:01 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 12:37:01 -0700
Subject: [Freeswitch-users] DialPlan Generation
In-Reply-To: <CAFvodxXx0nN8kfg5V-E7M3oTVC34Sr_RJ-EgxzULQELLyZ+6tA@mail.gmail.com>
References: <CAFvodxXx0nN8kfg5V-E7M3oTVC34Sr_RJ-EgxzULQELLyZ+6tA@mail.gmail.com>
Message-ID: <CAKzWOxU9Tkpe2JMB_gUb5P2UneScyTkHaxzpy33R_RFHdH=sRQ@mail.gmail.com>

On Thu, Aug 30, 2012 at 9:59 AM, Subhash <gvvsubhashkumar at gmail.com> wrote:

> Hi,
>
>      Which is the best way to create(static or dynamic) dialplan?
>
>     Please suggest me.
>
> The larger and more active your system becomes, the more valuable a
dynamic dialplan (and other configs) become. As to how you do the dynamic
stuff, that's really up to you. Using mod_xml_curl is a great way to make
it very scalable and have fallback options.
-MC
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From msc at freeswitch.org  Thu Aug 30 23:39:02 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 12:39:02 -0700
Subject: [Freeswitch-users] how to play a prompt after connection
In-Reply-To: <k1of31$hhj$1@ger.gmane.org>
References: <23F564DC-911A-4D5B-A63F-D420C755B216@beachdognet.com>
	<F62AB8C5-4184-4937-9AE0-27EF32F000ED@beachdognet.com>
	<k1nn6g$kd6$1@ger.gmane.org>
	<CAKzWOxW7HSWv5L28QenfCXbTsWYBSXkT+C0R+FNc5tMunuvdvw@mail.gmail.com>
	<k1od6m$ik$1@ger.gmane.org> <k1of31$hhj$1@ger.gmane.org>
Message-ID: <CAKzWOxW0okUosEJg6_mKJMfCSULV+hxa5v56DA2ryWWn-rWHKQ@mail.gmail.com>

Dave,
Can you turn off the sofia debugging? It's almost completely useless in
these scenarios. (only a few devs really understand it and they'll let you
know if you need to turn it on.) Also, please use "FreeSWITCH Log" as
syntax highlighting - that makes life much easier for those of us whose
eyes have higher mileage...
-MC

On Thu, Aug 30, 2012 at 12:31 PM, Dave Horton <daveh at beachdognet.com> wrote:

> On 2012-08-30 18:59:34 +0000, Dave Horton said:
>
> > Next I will try the conferencing version and paste those
> > results
>
> Below is the configuration I used with the bridged conferencing.
> Something must be wrong because there is no audio path between the a
> and b legs; also, when b hangs up a stays connected.
>
> Pastebin is at http://pastebin.freeswitch.org/19786 and configuration
> is below.  I'd welcome any thoughts on what is wrong...
>
>
>     <extension name="6174973896" continue="false">
>       <condition field="source" expression="mod_sofia"/>
>       <condition field="destination_number" expression="6174973896">
>                 <action application="set" data="call_limit=1"/>
>                 <action application="set" data="callFactor=1"/>
>                 <action application="set" data="customer=cps_play"/>
>                 <action application="ring_ready"/>
>                 <action application="pre_answer"/>
>                 <action application="start_dtmf"/>
>                 <action application="set"
> data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/>
>                 <action application="set"
> data="outfile=$${base_dir}/recordings/${call_id}.wav"/>
>                 <action application="export"
> data="nolocal:jitterbuffer_msec=100"/>
>                 <action application="set" data="RECORD_STEREO=true"/>
>                 <action application="set" data="aleg_uuid=${uuid}"/>
>                 <action application="set"
> data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/>
>                 <action application="set"
> data="outfile=$${base_dir}/recordings/${call_id}.wav"/>
>                 <action application="record_session" data="${outfile}"/>
>                 <action application="set" data="progress_timeout=60"/>
>                 <action application="set" data="call_timeout=120"/>
>                 <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE"/>
>                 <action application="set" data="hangup_after_bridge=true"/>
>                 <action application="conference"
> data="bridge:_uuid_ at simple:sofia/normal_customer/15083084809
> @${egress_gateway}"/>
>
>                 <action application="set"
> data="conference_enter_sound=misc/this-call-is-being-recorded"/>
>
>         </condition>
>     </extension>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From daveh at beachdognet.com  Fri Aug 31 00:05:20 2012
From: daveh at beachdognet.com (Dave Horton)
Date: Thu, 30 Aug 2012 16:05:20 -0400
Subject: [Freeswitch-users] how to play a prompt after connection
References: <23F564DC-911A-4D5B-A63F-D420C755B216@beachdognet.com>
	<F62AB8C5-4184-4937-9AE0-27EF32F000ED@beachdognet.com>
	<k1nn6g$kd6$1@ger.gmane.org>
	<CAKzWOxW7HSWv5L28QenfCXbTsWYBSXkT+C0R+FNc5tMunuvdvw@mail.gmail.com>
	<k1od6m$ik$1@ger.gmane.org> <k1of31$hhj$1@ger.gmane.org>
	<CAKzWOxW0okUosEJg6_mKJMfCSULV+hxa5v56DA2ryWWn-rWHKQ@mail.gmail.com>
Message-ID: <k1oh1v$2rs$1@ger.gmane.org>

OK, here is a pastebin minus the sip tracing with the highlighting: 
http://pastebin.freeswitch.org/19788

On 2012-08-30 19:39:02 +0000, Michael Collins said:

> Dave,
> Can you turn off the sofia debugging? It's almost completely useless in 
> these scenarios. (only a few devs really understand it and they'll let 
> you know if you need to turn it on.) Also, please use "FreeSWITCH Log" 
> as syntax highlighting - that makes life much easier for those of us 
> whose eyes have higher mileage...
> -MC
> 
> On Thu, Aug 30, 2012 at 12:31 PM, Dave Horton 
> <daveh at beachdognet.com> wrote:
> On 2012-08-30 18:59:34 +0000, Dave Horton said:
> 
> > Next I will try the conferencing version and paste those
> > results
> 
> Below is the configuration I used with the bridged conferencing.
> Something must be wrong because there is no audio path between the a
> and b legs; also, when b hangs up a stays connected.
> 
> Pastebin is at http://pastebin.freeswitch.org/19786 and configuration
> is below. ?I'd welcome any thoughts on what is wrong...
> 
> 
> ? ? <extension name="6174973896" continue="false">
> ? ? ? <condition field="source" expression="mod_sofia"/>
> ? ? ? <condition field="destination_number" expression="6174973896">
> ? ? ? ? ? ? ? ? <action application="set" data="call_limit=1"/>
> ? ? ? ? ? ? ? ? <action application="set" data="callFactor=1"/>
> ? ? ? ? ? ? ? ? <action application="set" data="customer=cps_play"/>
> ? ? ? ? ? ? ? ? <action application="ring_ready"/>
> ? ? ? ? ? ? ? ? <action application="pre_answer"/>
> ? ? ? ? ? ? ? ? <action application="start_dtmf"/>
> ? ? ? ? ? ? ? ? <action application="set"
> data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/>
> ? ? ? ? ? ? ? ? <action application="set"
> data="outfile=$${base_dir}/recordings/${call_id}.wav"/>
> ? ? ? ? ? ? ? ? <action application="export"
> data="nolocal:jitterbuffer_msec=100"/>
> ? ? ? ? ? ? ? ? <action application="set" data="RECORD_STEREO=true"/>
> ? ? ? ? ? ? ? ? <action application="set" data="aleg_uuid=${uuid}"/>
> ? ? ? ? ? ? ? ? <action application="set"
> data="call_id=${strftime(%Y%m%d_%H%M%S)}_${sip_from_tag}"/>
> ? ? ? ? ? ? ? ? <action application="set"
> data="outfile=$${base_dir}/recordings/${call_id}.wav"/>
> ? ? ? ? ? ? ? ? <action application="record_session" data="${outfile}"/>
> ? ? ? ? ? ? ? ? <action application="set" data="progress_timeout=60"/>
> ? ? ? ? ? ? ? ? <action application="set" data="call_timeout=120"/>
> ? ? ? ? ? ? ? ? <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE"/>
> ? ? ? ? ? ? ? ? <action application="set" data="hangup_after_bridge=true"/>
> ? ? ? ? ? ? ? ? <action application="conference"
> data="bridge:_uuid_ at simple:sofia/normal_customer/15083084809@${egress_gateway}"/> 
> 
> 
> ? ? ? ? ? ? ? ? <action application="set"
> data="conference_enter_sound=misc/this-call-is-being-recorded"/>
> 
> ? ? ? ? </condition>
> ? ? </extension>
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 freeswitch-users at vocalspace.com  Fri Aug 31 00:24:29 2012
From: freeswitch-users at vocalspace.com (Phillip Boles)
Date: Thu, 30 Aug 2012 15:24:29 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging in
	CS_SOFT_EXEC state
In-Reply-To: <FB1F0348-B48D-44FA-821D-1473E4EEDCE1@vocalspace.com>
References: <FB1F0348-B48D-44FA-821D-1473E4EEDCE1@vocalspace.com>
Message-ID: <5C0B6578-49A5-4174-87B4-C06E409EB341@vocalspace.com>

Sorry for the excessive logs. Here is my call to originate.
	var session =  ManagedSession.OriginateHandleHangup(s, dialString, ts, (y) => {
		try {
			Log.WriteLine(LogLevel.Info, "Hanging UP: "+ y.GetUuid());
			y.SetAutoHangup(true);
			y.destroy();

		} catch( Exception ) {
			Log.WriteLine(LogLevel.Critical, "Exception While Trying to handup");	
		}
	});

My hangup callback is getting hit and I am destroying the session

2012-08-28 10:49:27.296108 [INFO] switch_cpp.cpp:1227 Handing UP: e315f2e8-1fa8-4fd9-849b-f687dad8aed5
This is the only call on the system as it is a develpment machine and I see the call state being changed.
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:79 sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682 (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep
2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:407 (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY

If I call show channels after the above output it show there is a session sitting in CS_SOFT_EXEC corresponding to UUID e315f2e8-1fa8-4fd9-849b-f687dad8aed5.
Is there something else I need to do to release the lock on this session to let the resources be reclaimed.
Thanks!
Phillip


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From anthony.minessale at gmail.com  Fri Aug 31 00:48:07 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 30 Aug 2012 15:48:07 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging in
 CS_SOFT_EXEC state
In-Reply-To: <5C0B6578-49A5-4174-87B4-C06E409EB341@vocalspace.com>
References: <FB1F0348-B48D-44FA-821D-1473E4EEDCE1@vocalspace.com>
	<5C0B6578-49A5-4174-87B4-C06E409EB341@vocalspace.com>
Message-ID: <CAKbxfG88RmjT31OuXuQOczo5APhOOCLS7LhpmxKAh5myFTPxXw@mail.gmail.com>

destroy method should have a log line about (destroy/unlink session from object)
try calling your garbage collector, this is common issue with scripts
and make sure you are on latest GIT build


On Thu, Aug 30, 2012 at 3:24 PM, Phillip Boles
<freeswitch-users at vocalspace.com> wrote:
> Sorry for the excessive logs. Here is my call to originate.
>
> 	var session =  ManagedSession.OriginateHandleHangup(s, dialString, ts, (y)
> => {
> 		try {
> 			Log.WriteLine(LogLevel.Info, "Hanging UP: "+ y.GetUuid());
> 			y.SetAutoHangup(true);
> 			y.destroy();
>
> 		} catch( Exception ) {
> 			Log.WriteLine(LogLevel.Critical, "Exception While Trying to handup");	
> 		}
> 	});
>
>
> My hangup callback is getting hit and I am destroying the session
>
> 2012-08-28 10:49:27.296108 [INFO] switch_cpp.cpp:1227 Handing UP:
> e315f2e8-1fa8-4fd9-849b-f687dad8aed5
>
> This is the only call on the system as it is a develpment machine and I see
> the call state being changed.
>
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:79
> sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682
> (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:407
> (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY
>
>
> If I call show channels after the above output it show there is a session
> sitting in CS_SOFT_EXEC corresponding to UUID
> e315f2e8-1fa8-4fd9-849b-f687dad8aed5.
> Is there something else I need to do to release the lock on this session to
> let the resources be reclaimed.
>
> Thanks!
>
> Phillip
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From gavin.henry at gmail.com  Fri Aug 31 01:23:48 2012
From: gavin.henry at gmail.com (Gavin Henry)
Date: Thu, 30 Aug 2012 22:23:48 +0100
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
	<CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
	<CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
Message-ID: <CAA8_NKAnX97E0dV7tpakg+wO-TzWdKm5wrm0qNP1EqODAVRoNw@mail.gmail.com>

> Centos5 worked well for me, Centos6 worked for crap... I went to debian 6
> (squeeze) and problems went a way... using the latest stable on the debian6
> line installing from the netinstall (non-free variant as I have hardware
> thats not supported under the main netinstall iso) and then installing
> freeswitch via the instructions on the wiki and no problems so far

Same here. We're CentOS 5 here and Debian Squeezy for new boxes.

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


From reza at lethalnetworks.com  Fri Aug 31 01:54:34 2012
From: reza at lethalnetworks.com (reza a)
Date: Thu, 30 Aug 2012 14:54:34 -0700 (PDT)
Subject: [Freeswitch-users] testing
Message-ID: <101795878.2521.1346363673997.JavaMail.root@lethalnetworks.com>

testing
dont know if i subscribed to this properly.


From grcamauer at gmail.com  Fri Aug 31 02:12:37 2012
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Thu, 30 Aug 2012 19:12:37 -0300
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAA8_NKAnX97E0dV7tpakg+wO-TzWdKm5wrm0qNP1EqODAVRoNw@mail.gmail.com>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
	<CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
	<CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
	<CAA8_NKAnX97E0dV7tpakg+wO-TzWdKm5wrm0qNP1EqODAVRoNw@mail.gmail.com>
Message-ID: <CAPNTnnpHzcUzJ8Jnkj6Nma_2pZVyVhfWDA4RjBtJkuNWEMOdvg@mail.gmail.com>

Is anyone having the same problems with RedHat or is this CentOS specific?


On Thu, Aug 30, 2012 at 6:23 PM, Gavin Henry <gavin.henry at gmail.com> wrote:

> > Centos5 worked well for me, Centos6 worked for crap... I went to debian 6
> > (squeeze) and problems went a way... using the latest stable on the
> debian6
> > line installing from the netinstall (non-free variant as I have hardware
> > thats not supported under the main netinstall iso) and then installing
> > freeswitch via the instructions on the wiki and no problems so far
>
> Same here. We're CentOS 5 here and Debian Squeezy for new boxes.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.surevoip.co.uk
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Guillermo Ruiz Camauer
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From msc at freeswitch.org  Fri Aug 31 02:16:08 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 15:16:08 -0700
Subject: [Freeswitch-users] testing
In-Reply-To: <101795878.2521.1346363673997.JavaMail.root@lethalnetworks.com>
References: <101795878.2521.1346363673997.JavaMail.root@lethalnetworks.com>
Message-ID: <CAKzWOxW-zS-8LChKLbsO6+=XLKvVCCrqMSy0XgjM5qS0=khggg@mail.gmail.com>

On Thu, Aug 30, 2012 at 2:54 PM, reza a <reza at lethalnetworks.com> wrote:

> testing
> dont know if i subscribed to this properly.
>
Well, you certainly posted successfully! :)
-MC
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From msc at freeswitch.org  Fri Aug 31 02:17:03 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 15:17:03 -0700
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAPNTnnpHzcUzJ8Jnkj6Nma_2pZVyVhfWDA4RjBtJkuNWEMOdvg@mail.gmail.com>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
	<CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
	<CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
	<CAA8_NKAnX97E0dV7tpakg+wO-TzWdKm5wrm0qNP1EqODAVRoNw@mail.gmail.com>
	<CAPNTnnpHzcUzJ8Jnkj6Nma_2pZVyVhfWDA4RjBtJkuNWEMOdvg@mail.gmail.com>
Message-ID: <CAKzWOxVCk=tzE9tM+LDHW0pLjzJPQxW7npdM8JQhi6rAe8M2Xg@mail.gmail.com>

On Thu, Aug 30, 2012 at 3:12 PM, Guillermo Ruiz Camauer <grcamauer at gmail.com
> wrote:

> Is anyone having the same problems with RedHat or is this CentOS specific?
>
> An excellent question. I'd really like to know. Same with those running
Scientific Linux (SL) - it's RHEL based also.
-MC
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From cmason at frontiernetworks.ca  Fri Aug 31 02:17:12 2012
From: cmason at frontiernetworks.ca (Colin Mason)
Date: Thu, 30 Aug 2012 18:17:12 -0400
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAPNTnnpHzcUzJ8Jnkj6Nma_2pZVyVhfWDA4RjBtJkuNWEMOdvg@mail.gmail.com>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
	<CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
	<CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
	<CAA8_NKAnX97E0dV7tpakg+wO-TzWdKm5wrm0qNP1EqODAVRoNw@mail.gmail.com>
	<CAPNTnnpHzcUzJ8Jnkj6Nma_2pZVyVhfWDA4RjBtJkuNWEMOdvg@mail.gmail.com>
Message-ID: <0D1C698866F66045A6201FD0F59CAC90014678ACA7@EX.frontier.local>

What exactly is the problem?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Guillermo Ruiz Camauer
Sent: Thursday, August 30, 2012 6:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] RedHat 6.X Performance

Is anyone having the same problems with RedHat or is this CentOS specific?

On Thu, Aug 30, 2012 at 6:23 PM, Gavin Henry <gavin.henry at gmail.com<mailto:gavin.henry at gmail.com>> wrote:
> Centos5 worked well for me, Centos6 worked for crap... I went to debian 6
> (squeeze) and problems went a way... using the latest stable on the debian6
> line installing from the netinstall (non-free variant as I have hardware
> thats not supported under the main netinstall iso) and then installing
> freeswitch via the instructions on the wiki and no problems so far
Same here. We're CentOS 5 here and Debian Squeezy for new boxes.

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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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



--
Guillermo Ruiz Camauer
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From drk at drkngs.net  Fri Aug 31 02:18:39 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Thu, 30 Aug 2012 15:18:39 -0700
Subject: [Freeswitch-users]
	=?iso-8859-1?q?Problem_with_originated_calls_h?=
	=?iso-8859-1?q?anging_in=09CS=5FSOFT=5FEXEC_state?=
In-Reply-To: <FB1F0348-B48D-44FA-821D-1473E4EEDCE1@vocalspace.com>
Message-ID: <20120830221839.7f6cf44b@mail.tritonwest.net>

Argh, I wish someone with access to the wiki would fix this.......  
   
The old methods for making outbound legs from mod_managed, are just that, old. Where are you calling this from? If you want to originate a call from managed code the only way you won't run into some problem with cross_app_domain stuff, is to use Api.Execute("originate",...), and if you want to bridge a call then use Session.Execute("Bridge...").   
   
If you stick to this, you should have no problem. If you need a managed session handle when you do an originate, then have the Originate API call a managed dialplan app, and it can do the rest.  
   
--Dave
      _____  

  From: Phillip Boles [mailto:freeswitch-users at vocalspace.com]
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, 30 Aug 2012 11:20:06 -0700
Subject: [Freeswitch-users] Problem with originated calls hanging in CS_SOFT_EXEC state



    var session =  ManagedSession.OriginateHandleHangup(s, dialString, ts, (y) => {    try {     Log.WriteLine(LogLevel.Info, "Hanging UP: "+ y.GetUuid());     y.SetAutoHangup(true);     y.destroy();      } catch( Exception ) {     Log.WriteLine(LogLevel.Critical, "Exception While Trying to handup");     }   });     

   2012-08-28 10:49:12.156172 [INFO] switch_cpp.cpp:1227 Dialing  2012-08-28 10:49:12.156172 [DEBUG] switch_ivr_originate.c:1947 Parsing global variables  2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [ignore_early_media]=[true]  2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [origination_caller_id_number]=[2223334444]  2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [origination_caller_id_name]=[Vocalspace]  2012-08-28 10:49:12.156172 [NOTICE] switch_channel.c:926 New Channel sofia/external/XXXXXXXXXXX [e315f2e8-1fa8-4fd9-849b-f687dad8aed5]  2012-08-28 10:49:12.156172 [DEBUG] mod_sofia.c:4709 (sofia/external/XXXXXXXXXXX) State Change CS_NEW -> CS_INIT  2012-08-28 10:49:12.156172 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:12.156172 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_INIT  2012-08-28 10:49:12.156172 [DEBUG] switch_core_state_machine.c:421 (sofia/external/XXXXXXXXXXX) State INIT  2012-08-28 10:49:12.156172 [DEBUG] mod_sofia.c:85 sofia/external/XXXXXXXXXXX SOFIA INIT  2012-08-28 10:49:12.216161 [DEBUG] switch_nat.c:511 mapped public port 19628 protocol UDP to localport 19628  2012-08-28 10:49:12.256159 [DEBUG] switch_nat.c:511 mapped public port 19629 protocol UDP to localport 19629  2012-08-28 10:49:12.256159 [DEBUG] mod_sofia.c:129 (sofia/external/XXXXXXXXXXX) State Change CS_INIT -> CS_ROUTING  2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:421 (sofia/external/XXXXXXXXXXX) State INIT going to sleep  2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_ROUTING  2012-08-28 10:49:12.256159 [DEBUG] switch_channel.c:1887 (sofia/external/XXXXXXXXXXX) Callstate Change DOWN -> RINGING  2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:430 (sofia/external/XXXXXXXXXXX) State ROUTING  2012-08-28 10:49:12.256159 [DEBUG] mod_sofia.c:152 sofia/external/XXXXXXXXXXX SOFIA ROUTING  2012-08-28 10:49:12.256159 [DEBUG] switch_ivr_originate.c:67 (sofia/external/XXXXXXXXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA  2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:430 (sofia/external/XXXXXXXXXXX) State ROUTING going to sleep  2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_CONSUME_MEDIA  2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:449 (sofia/external/XXXXXXXXXXX) State CONSUME_MEDIA  2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:449 (sofia/external/XXXXXXXXXXX) State CONSUME_MEDIA going to sleep  2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:12.256159 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [calling][0]  2012-08-28 10:49:12.636131 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:12.636131 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:12.636131 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call"   2012-08-28 10:49:12.636131 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [proceeding][180]  2012-08-28 10:49:12.636131 [NOTICE] sofia.c:5769 Ring-Ready sofia/external/XXXXXXXXXXX!  2012-08-28 10:49:14.655990 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:14.655990 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:14.655990 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call"   2012-08-28 10:49:14.655990 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [proceeding][183]  2012-08-28 10:49:14.655990 [DEBUG] sofia.c:5688 Remote SDP:  v=0  o=FreeSWITCH 1346150407 1346150408 IN IP4 8.19.97.6  s=FreeSWITCH  c=IN IP4 8.19.97.6  t=0 0  m=audio 18534 RTP/AVP 0 101 13  a=rtpmap:0 PCMU/8000  a=rtpmap:101 telephone-event/8000  a=fmtp:101 0-16  a=rtpmap:13 CN/8000  a=ptime:20    2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:4911 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]  2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:2995 Set Codec sofia/external/XXXXXXXXXXX PCMU/8000 20 ms 160 samples 64000 bits  2012-08-28 10:49:14.655990 [DEBUG] switch_core_codec.c:111 sofia/external/XXXXXXXXXXX Original read codec set to PCMU:0  2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:5025 Set 2833 dtmf send payload to 101  2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3244 AUDIO RTP [sofia/external/XXXXXXXXXXX] 10.10.80.33 port 19628 -> 8.19.97.6 port 18534 codec: 0 ms: 20  2012-08-28 10:49:14.655990 [DEBUG] switch_rtp.c:1676 Starting timer [soft] 160 bytes per 20ms  2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3508 Set 2833 dtmf send payload to 101  2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3514 Set 2833 dtmf receive payload to 101  2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3541 sofia/external/XXXXXXXXXXX Set rtp dtmf delay to 40  2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3547 Set comfort noise payload to 13  2012-08-28 10:49:14.655990 [NOTICE] sofia_glue.c:4052 Pre-Answer sofia/external/XXXXXXXXXXX!  2012-08-28 10:49:14.655990 [DEBUG] switch_channel.c:2986 (sofia/external/XXXXXXXXXXX) Callstate Change RINGING -> EARLY  2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:22.136467 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call"   2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [completing][200]  2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5685 Duplicate SDP  v=0  o=FreeSWITCH 1346150407 1346150408 IN IP4 8.19.97.6  s=FreeSWITCH  c=IN IP4 8.19.97.6  t=0 0  m=audio 18534 RTP/AVP 0 101 13  a=rtpmap:0 PCMU/8000  a=rtpmap:101 telephone-event/8000  a=fmtp:101 0-16  a=rtpmap:13 CN/8000  a=ptime:20    2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [ready][200]  2012-08-28 10:49:22.136467 [DEBUG] switch_channel.c:3245 (sofia/external/XXXXXXXXXXX) Callstate Change EARLY -> ACTIVE  2012-08-28 10:49:22.136467 [NOTICE] sofia.c:6352 Channel [sofia/external/XXXXXXXXXXX] has been answered  2012-08-28 10:49:22.136467 [DEBUG] switch_ivr_originate.c:3330 Originate Resulted in Success: [sofia/external/XXXXXXXXXXX]  2012-08-28 10:49:22.136467 [DEBUG] switch_cpp.cpp:1083 (sofia/external/XXXXXXXXXXX) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE  2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_SOFT_EXECUTE  2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:443 (sofia/external/XXXXXXXXXXX) State SOFT_EXECUTE  2012-08-28 10:49:22.136467 [DEBUG] mod_sofia.c:592 SOFIA SOFT_EXECUTE  2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:261 sofia/external/XXXXXXXXXXX Standard SOFT_EXECUTE  2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:443 (sofia/external/XXXXXXXXXXX) State SOFT_EXECUTE going to sleep  2012-08-28 10:49:22.136467 [DEBUG] switch_cpp.cpp:1172 CoreSession::seHangupHook, hangup_func: (nil)  2012-08-28 10:49:22.136467 [INFO] switch_cpp.cpp:1227 Connected  2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Session Ready  2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Call Initialized  2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Trying To Run  2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Running  2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Checking for file /svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw  2012-08-28 10:49:23.160407 [ERR] mod_sndfile.c:197 Error Opening File [/svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw] [System error : No such file or directory.]  2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Transcoding xfile /svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw  2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Sending MQ Message  2012-08-28 10:49:23.296392 [NOTICE] switch_cpp.cpp:1227 Creating a channel...  2012-08-28 10:49:23.296392 [NOTICE] switch_cpp.cpp:1227 Defining Exchange FS...  2012-08-28 10:49:23.316389 [NOTICE] switch_cpp.cpp:1227 Binding Exchange and Queue..  2012-08-28 10:49:23.336389 [NOTICE] switch_cpp.cpp:1227 Sending message..  2012-08-28 10:49:23.356388 [NOTICE] switch_cpp.cpp:1227 shout://www.product.com/files/media/e53bc5bf-4e00-ab8d-4e4d-b201699628d1.mp3  2012-08-28 10:49:25.916204 [DEBUG] mod_shout.c:472 Read Thread Done  2012-08-28 10:49:26.356173 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms  2012-08-28 10:49:26.596156 [DEBUG] switch_rtp.c:3252 Correct ip/port confirmed.  2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:27.296108 [DEBUG] switch_channel.c:2849 (sofia/external/XXXXXXXXXXX) Callstate Change ACTIVE -> HANGUP  2012-08-28 10:49:27.296108 [NOTICE] sofia.c:694 Hangup sofia/external/XXXXXXXXXXX [CS_SOFT_EXECUTE] [NORMAL_CLEARING]  2012-08-28 10:49:27.296108 [DEBUG] switch_channel.c:2872 Send signal sofia/external/XXXXXXXXXXX [KILL]  2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1227 AppFunction is in hangupCallback.  2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_HANGUP  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:622 (sofia/external/XXXXXXXXXXX) State HANGUP  2012-08-28 10:49:27.296108 [DEBUG] mod_sofia.c:473 Channel sofia/external/XXXXXXXXXXX hanging up, cause: NORMAL_CLEARING  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:47 sofia/external/XXXXXXXXXXX Standard HANGUP, cause: NORMAL_CLEARING  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:622 (sofia/external/XXXXXXXXXXX) State HANGUP going to sleep  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:413 (sofia/external/XXXXXXXXXXX) State Change CS_HANGUP -> CS_REPORTING  2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1227 AppFunction is in hangupCallback.  2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_REPORTING  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682 (sofia/external/XXXXXXXXXXX) State REPORTING  2012-08-28 10:49:27.296108 [INFO] switch_cpp.cpp:1227 Handing UP: e315f2e8-1fa8-4fd9-849b-f687dad8aed5  2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1007 sofia/external/XXXXXXXXXXX destroy/unlink session from object  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:79 sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682 (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:407 (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY  2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1424 Session 4 (sofia/external/XXXXXXXXXXX) Locked, Waiting on external entities  2012-08-28 10:49:27.316110 [DEBUG] switch_ivr_play_say.c:1682 done playing file shout://www.product.com/files/media/e53bc5bf-4e00-ab8d-4e4d-b201699628d1.mp3    freeswitch at fs03.int.colo> show channels    uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num  e315f2e8-1fa8-4fd9-849b-f687dad8aed5,outbound,2012-08-28 10:49:12,1346168952,sofia/external/XXXXXXXXXXX,CS_SOFT_EXECUTE,Vocalspace,2223334444,,XXXXXXXXXXX,,,,default,PCMU,8000,64000,PCMU,8000,64000,,fs03.int.colo,,,ACTIVE,Outbound Call,XXXXXXXXXXX,,,,    1 total.    2012-08-28 10:58:10.256585 [DEBUG] switch_nat.c:511 mapped public port 5060 protocol TCP to localport 5060  2012-08-28 10:58:10.296583 [DEBUG] switch_nat.c:511 mapped public port 5080 protocol UDP to localport 5080  2012-08-28 10:58:10.336580 [DEBUG] switch_nat.c:511 mapped public port 5080 protocol TCP to localport 5080  2012-08-28 10:58:10.396576 [DEBUG] switch_nat.c:511 mapped public port 19628 protocol UDP to localport 19628  2012-08-28 10:58:10.436573 [DEBUG] switch_nat.c:511 mapped public port 19629 protocol UDP to localport 19629  2012-08-28 10:58:10.476570 [DEBUG] switch_nat.c:511 mapped public port 22013 protocol UDP to localport 22013  2012-08-28 10:58:10.536566 [DEBUG] switch_nat.c:511 mapped public port 22191 protocol UDP to localport 22191  2012-08-28 10:58:10.576563 [DEBUG] switch_nat.c:511 mapped public port 25326 protocol UDP to localport 25326  2012-08-28 10:58:10.636559 [DEBUG] switch_nat.c:511 mapped public port 25327 protocol UDP to localport 25327  Loading /usr/lib64/freeswitch/mod/managed/Enyim.Caching.dll from domain Enyim.Caching.dll_63  2012-08-28 11:00:02.097780 [DEBUG] switch_cpp.cpp:1227 Assembly Enyim.Caching, Version=1.2.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/Jint.dll from domain Jint.dll_64  2012-08-28 11:00:02.316760 [DEBUG] switch_cpp.cpp:1227 Assembly Jint, Version=0.9.2.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/log4net.dll from domain log4net.dll_65  2012-08-28 11:00:02.536745 [DEBUG] switch_cpp.cpp:1227 Assembly log4net, Version=1.2.10.0, Culture=neutral, PublicKeyToken=1b44e1d426115821 doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/Antlr3.Runtime.dll from domain Antlr3.Runtime.dll_66  2012-08-28 11:00:02.756720 [DEBUG] switch_cpp.cpp:1227 Assembly Antlr3.Runtime, Version=3.3.1.7705, Culture=neutral, PublicKeyToken=eb42632606e9261f doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/RabbitMQ.Client.dll from domain RabbitMQ.Client.dll_67  2012-08-28 11:00:02.996708 [DEBUG] switch_cpp.cpp:1227 Assembly RabbitMQ.Client, Version=0.0.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/ManagedEsl.dll from domain ManagedEsl.dll_68  2012-08-28 11:00:03.216689 [DEBUG] switch_cpp.cpp:1227 Assembly ManagedEsl, Version=1.0.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/FreeSWITCH.Managed.dll from domain FreeSWITCH.Managed.dll_69  2012-08-28 11:00:03.436678 [DEBUG] switch_cpp.cpp:1227 Assembly FreeSWITCH.Managed, Version=1.0.5.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/FreeSwitch.EventSocket.dll from domain FreeSwitch.EventSocket.dll_70  2012-08-28 11:00:03.657671 [DEBUG] switch_cpp.cpp:1227 Assembly FreeSwitch.EventSocket, Version=1.0.0.0, Culture=neutral, PublicKeyToken=94d8a22cdb4b7713 doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/Newtonsoft.Json.dll from domain Newtonsoft.Json.dll_71  2012-08-28 11:00:03.876655 [DEBUG] switch_cpp.cpp:1227 Assembly Newtonsoft.Json, Version=4.5.0.0, Culture=neutral, PublicKeyToken=30ad4fe6b2a6aeed doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/Freeswitch.UUID.dll from domain Freeswitch.UUID.dll_72  2012-08-28 11:00:04.096628 [DEBUG] switch_cpp.cpp:1227 Assembly Freeswitch.UUID, Version=1.0.0.0, Culture=neutral, PublicKeyToken=f758751257b2cdaf doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/Product.dll from domain Product.dll_73    freeswitch at fs03.int.colo> show channels      0 total.    

    var session =  ManagedSession.OriginateHandleHangup(s, dialString, ts, (y) => {    try {     Log.WriteLine(LogLevel.Info, "Hanging UP: "+ y.GetUuid());     y.SetAutoHangup(true);     y.destroy();      } catch( Exception ) {     Log.WriteLine(LogLevel.Critical, "Exception While Trying to handup");     }   });     

   2012-08-28 10:49:12.156172 [INFO] switch_cpp.cpp:1227 Dialing  2012-08-28 10:49:12.156172 [DEBUG] switch_ivr_originate.c:1947 Parsing global variables  2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [ignore_early_media]=[true]  2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [origination_caller_id_number]=[2223334444]  2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [origination_caller_id_name]=[Vocalspace]  2012-08-28 10:49:12.156172 [NOTICE] switch_channel.c:926 New Channel sofia/external/XXXXXXXXXXX [e315f2e8-1fa8-4fd9-849b-f687dad8aed5]  2012-08-28 10:49:12.156172 [DEBUG] mod_sofia.c:4709 (sofia/external/XXXXXXXXXXX) State Change CS_NEW -> CS_INIT  2012-08-28 10:49:12.156172 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:12.156172 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_INIT  2012-08-28 10:49:12.156172 [DEBUG] switch_core_state_machine.c:421 (sofia/external/XXXXXXXXXXX) State INIT  2012-08-28 10:49:12.156172 [DEBUG] mod_sofia.c:85 sofia/external/XXXXXXXXXXX SOFIA INIT  2012-08-28 10:49:12.216161 [DEBUG] switch_nat.c:511 mapped public port 19628 protocol UDP to localport 19628  2012-08-28 10:49:12.256159 [DEBUG] switch_nat.c:511 mapped public port 19629 protocol UDP to localport 19629  2012-08-28 10:49:12.256159 [DEBUG] mod_sofia.c:129 (sofia/external/XXXXXXXXXXX) State Change CS_INIT -> CS_ROUTING  2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:421 (sofia/external/XXXXXXXXXXX) State INIT going to sleep  2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_ROUTING  2012-08-28 10:49:12.256159 [DEBUG] switch_channel.c:1887 (sofia/external/XXXXXXXXXXX) Callstate Change DOWN -> RINGING  2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:430 (sofia/external/XXXXXXXXXXX) State ROUTING  2012-08-28 10:49:12.256159 [DEBUG] mod_sofia.c:152 sofia/external/XXXXXXXXXXX SOFIA ROUTING  2012-08-28 10:49:12.256159 [DEBUG] switch_ivr_originate.c:67 (sofia/external/XXXXXXXXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA  2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:430 (sofia/external/XXXXXXXXXXX) State ROUTING going to sleep  2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_CONSUME_MEDIA  2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:449 (sofia/external/XXXXXXXXXXX) State CONSUME_MEDIA  2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:449 (sofia/external/XXXXXXXXXXX) State CONSUME_MEDIA going to sleep  2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:12.256159 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [calling][0]  2012-08-28 10:49:12.636131 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:12.636131 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:12.636131 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call"   2012-08-28 10:49:12.636131 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [proceeding][180]  2012-08-28 10:49:12.636131 [NOTICE] sofia.c:5769 Ring-Ready sofia/external/XXXXXXXXXXX!  2012-08-28 10:49:14.655990 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:14.655990 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:14.655990 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call"   2012-08-28 10:49:14.655990 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [proceeding][183]  2012-08-28 10:49:14.655990 [DEBUG] sofia.c:5688 Remote SDP:  v=0  o=FreeSWITCH 1346150407 1346150408 IN IP4 8.19.97.6  s=FreeSWITCH  c=IN IP4 8.19.97.6  t=0 0  m=audio 18534 RTP/AVP 0 101 13  a=rtpmap:0 PCMU/8000  a=rtpmap:101 telephone-event/8000  a=fmtp:101 0-16  a=rtpmap:13 CN/8000  a=ptime:20    2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:4911 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]  2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:2995 Set Codec sofia/external/XXXXXXXXXXX PCMU/8000 20 ms 160 samples 64000 bits  2012-08-28 10:49:14.655990 [DEBUG] switch_core_codec.c:111 sofia/external/XXXXXXXXXXX Original read codec set to PCMU:0  2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:5025 Set 2833 dtmf send payload to 101  2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3244 AUDIO RTP [sofia/external/XXXXXXXXXXX] 10.10.80.33 port 19628 -> 8.19.97.6 port 18534 codec: 0 ms: 20  2012-08-28 10:49:14.655990 [DEBUG] switch_rtp.c:1676 Starting timer [soft] 160 bytes per 20ms  2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3508 Set 2833 dtmf send payload to 101  2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3514 Set 2833 dtmf receive payload to 101  2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3541 sofia/external/XXXXXXXXXXX Set rtp dtmf delay to 40  2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3547 Set comfort noise payload to 13  2012-08-28 10:49:14.655990 [NOTICE] sofia_glue.c:4052 Pre-Answer sofia/external/XXXXXXXXXXX!  2012-08-28 10:49:14.655990 [DEBUG] switch_channel.c:2986 (sofia/external/XXXXXXXXXXX) Callstate Change RINGING -> EARLY  2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:22.136467 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call"   2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [completing][200]  2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5685 Duplicate SDP  v=0  o=FreeSWITCH 1346150407 1346150408 IN IP4 8.19.97.6  s=FreeSWITCH  c=IN IP4 8.19.97.6  t=0 0  m=audio 18534 RTP/AVP 0 101 13  a=rtpmap:0 PCMU/8000  a=rtpmap:101 telephone-event/8000  a=fmtp:101 0-16  a=rtpmap:13 CN/8000  a=ptime:20    2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [ready][200]  2012-08-28 10:49:22.136467 [DEBUG] switch_channel.c:3245 (sofia/external/XXXXXXXXXXX) Callstate Change EARLY -> ACTIVE  2012-08-28 10:49:22.136467 [NOTICE] sofia.c:6352 Channel [sofia/external/XXXXXXXXXXX] has been answered  2012-08-28 10:49:22.136467 [DEBUG] switch_ivr_originate.c:3330 Originate Resulted in Success: [sofia/external/XXXXXXXXXXX]  2012-08-28 10:49:22.136467 [DEBUG] switch_cpp.cpp:1083 (sofia/external/XXXXXXXXXXX) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE  2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_SOFT_EXECUTE  2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:443 (sofia/external/XXXXXXXXXXX) State SOFT_EXECUTE  2012-08-28 10:49:22.136467 [DEBUG] mod_sofia.c:592 SOFIA SOFT_EXECUTE  2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:261 sofia/external/XXXXXXXXXXX Standard SOFT_EXECUTE  2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:443 (sofia/external/XXXXXXXXXXX) State SOFT_EXECUTE going to sleep  2012-08-28 10:49:22.136467 [DEBUG] switch_cpp.cpp:1172 CoreSession::seHangupHook, hangup_func: (nil)  2012-08-28 10:49:22.136467 [INFO] switch_cpp.cpp:1227 Connected  2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Session Ready  2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Call Initialized  2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Trying To Run  2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Running  2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Checking for file /svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw  2012-08-28 10:49:23.160407 [ERR] mod_sndfile.c:197 Error Opening File [/svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw] [System error : No such file or directory.]  2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Transcoding xfile /svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw  2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Sending MQ Message  2012-08-28 10:49:23.296392 [NOTICE] switch_cpp.cpp:1227 Creating a channel...  2012-08-28 10:49:23.296392 [NOTICE] switch_cpp.cpp:1227 Defining Exchange FS...  2012-08-28 10:49:23.316389 [NOTICE] switch_cpp.cpp:1227 Binding Exchange and Queue..  2012-08-28 10:49:23.336389 [NOTICE] switch_cpp.cpp:1227 Sending message..  2012-08-28 10:49:23.356388 [NOTICE] switch_cpp.cpp:1227 shout://www.product.com/files/media/e53bc5bf-4e00-ab8d-4e4d-b201699628d1.mp3  2012-08-28 10:49:25.916204 [DEBUG] mod_shout.c:472 Read Thread Done  2012-08-28 10:49:26.356173 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms  2012-08-28 10:49:26.596156 [DEBUG] switch_rtp.c:3252 Correct ip/port confirmed.  2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:27.296108 [DEBUG] switch_channel.c:2849 (sofia/external/XXXXXXXXXXX) Callstate Change ACTIVE -> HANGUP  2012-08-28 10:49:27.296108 [NOTICE] sofia.c:694 Hangup sofia/external/XXXXXXXXXXX [CS_SOFT_EXECUTE] [NORMAL_CLEARING]  2012-08-28 10:49:27.296108 [DEBUG] switch_channel.c:2872 Send signal sofia/external/XXXXXXXXXXX [KILL]  2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1227 AppFunction is in hangupCallback.  2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_HANGUP  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:622 (sofia/external/XXXXXXXXXXX) State HANGUP  2012-08-28 10:49:27.296108 [DEBUG] mod_sofia.c:473 Channel sofia/external/XXXXXXXXXXX hanging up, cause: NORMAL_CLEARING  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:47 sofia/external/XXXXXXXXXXX Standard HANGUP, cause: NORMAL_CLEARING  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:622 (sofia/external/XXXXXXXXXXX) State HANGUP going to sleep  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:413 (sofia/external/XXXXXXXXXXX) State Change CS_HANGUP -> CS_REPORTING  2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1227 AppFunction is in hangupCallback.  2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_REPORTING  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682 (sofia/external/XXXXXXXXXXX) State REPORTING  2012-08-28 10:49:27.296108 [INFO] switch_cpp.cpp:1227 Handing UP: e315f2e8-1fa8-4fd9-849b-f687dad8aed5  2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1007 sofia/external/XXXXXXXXXXX destroy/unlink session from object  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:79 sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682 (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep  2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:407 (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY  2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]  2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1424 Session 4 (sofia/external/XXXXXXXXXXX) Locked, Waiting on external entities  2012-08-28 10:49:27.316110 [DEBUG] switch_ivr_play_say.c:1682 done playing file shout://www.product.com/files/media/e53bc5bf-4e00-ab8d-4e4d-b201699628d1.mp3    freeswitch at fs03.int.colo> show channels    uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num  e315f2e8-1fa8-4fd9-849b-f687dad8aed5,outbound,2012-08-28 10:49:12,1346168952,sofia/external/XXXXXXXXXXX,CS_SOFT_EXECUTE,Vocalspace,2223334444,,XXXXXXXXXXX,,,,default,PCMU,8000,64000,PCMU,8000,64000,,fs03.int.colo,,,ACTIVE,Outbound Call,XXXXXXXXXXX,,,,    1 total.    2012-08-28 10:58:10.256585 [DEBUG] switch_nat.c:511 mapped public port 5060 protocol TCP to localport 5060  2012-08-28 10:58:10.296583 [DEBUG] switch_nat.c:511 mapped public port 5080 protocol UDP to localport 5080  2012-08-28 10:58:10.336580 [DEBUG] switch_nat.c:511 mapped public port 5080 protocol TCP to localport 5080  2012-08-28 10:58:10.396576 [DEBUG] switch_nat.c:511 mapped public port 19628 protocol UDP to localport 19628  2012-08-28 10:58:10.436573 [DEBUG] switch_nat.c:511 mapped public port 19629 protocol UDP to localport 19629  2012-08-28 10:58:10.476570 [DEBUG] switch_nat.c:511 mapped public port 22013 protocol UDP to localport 22013  2012-08-28 10:58:10.536566 [DEBUG] switch_nat.c:511 mapped public port 22191 protocol UDP to localport 22191  2012-08-28 10:58:10.576563 [DEBUG] switch_nat.c:511 mapped public port 25326 protocol UDP to localport 25326  2012-08-28 10:58:10.636559 [DEBUG] switch_nat.c:511 mapped public port 25327 protocol UDP to localport 25327  Loading /usr/lib64/freeswitch/mod/managed/Enyim.Caching.dll from domain Enyim.Caching.dll_63  2012-08-28 11:00:02.097780 [DEBUG] switch_cpp.cpp:1227 Assembly Enyim.Caching, Version=1.2.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/Jint.dll from domain Jint.dll_64  2012-08-28 11:00:02.316760 [DEBUG] switch_cpp.cpp:1227 Assembly Jint, Version=0.9.2.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/log4net.dll from domain log4net.dll_65  2012-08-28 11:00:02.536745 [DEBUG] switch_cpp.cpp:1227 Assembly log4net, Version=1.2.10.0, Culture=neutral, PublicKeyToken=1b44e1d426115821 doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/Antlr3.Runtime.dll from domain Antlr3.Runtime.dll_66  2012-08-28 11:00:02.756720 [DEBUG] switch_cpp.cpp:1227 Assembly Antlr3.Runtime, Version=3.3.1.7705, Culture=neutral, PublicKeyToken=eb42632606e9261f doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/RabbitMQ.Client.dll from domain RabbitMQ.Client.dll_67  2012-08-28 11:00:02.996708 [DEBUG] switch_cpp.cpp:1227 Assembly RabbitMQ.Client, Version=0.0.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/ManagedEsl.dll from domain ManagedEsl.dll_68  2012-08-28 11:00:03.216689 [DEBUG] switch_cpp.cpp:1227 Assembly ManagedEsl, Version=1.0.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/FreeSWITCH.Managed.dll from domain FreeSWITCH.Managed.dll_69  2012-08-28 11:00:03.436678 [DEBUG] switch_cpp.cpp:1227 Assembly FreeSWITCH.Managed, Version=1.0.5.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/FreeSwitch.EventSocket.dll from domain FreeSwitch.EventSocket.dll_70  2012-08-28 11:00:03.657671 [DEBUG] switch_cpp.cpp:1227 Assembly FreeSwitch.EventSocket, Version=1.0.0.0, Culture=neutral, PublicKeyToken=94d8a22cdb4b7713 doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/Newtonsoft.Json.dll from domain Newtonsoft.Json.dll_71  2012-08-28 11:00:03.876655 [DEBUG] switch_cpp.cpp:1227 Assembly Newtonsoft.Json, Version=4.5.0.0, Culture=neutral, PublicKeyToken=30ad4fe6b2a6aeed doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/Freeswitch.UUID.dll from domain Freeswitch.UUID.dll_72  2012-08-28 11:00:04.096628 [DEBUG] switch_cpp.cpp:1227 Assembly Freeswitch.UUID, Version=1.0.0.0, Culture=neutral, PublicKeyToken=f758751257b2cdaf doesn't reference FreeSWITCH.Managed, not loading.  Loading /usr/lib64/freeswitch/mod/managed/Product.dll from domain Product.dll_73    freeswitch at fs03.int.colo> show channels      0 total.          
   
 
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From drk at drkngs.net  Fri Aug 31 02:22:00 2012
From: drk at drkngs.net (Dave R. Kompel)
Date: Thu, 30 Aug 2012 15:22:00 -0700
Subject: [Freeswitch-users] Problem with originated calls hanging in
 CS_SOFT_EXEC state
In-Reply-To: <CAKbxfG88RmjT31OuXuQOczo5APhOOCLS7LhpmxKAh5myFTPxXw@mail.gmail.com>
Message-ID: <20120830222200.ff3a5e72@mail.tritonwest.net>

Actually, all the managed objects are derived from IDisposable, so you should use the .Dispose() method, and let the wrapper do it's job. 
      _____  

  From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Thu, 30 Aug 2012 13:48:07 -0700
Subject: Re: [Freeswitch-users] Problem with originated calls hanging in CS_SOFT_EXEC state

destroy method should have a log line about (destroy/unlink session from object)
try calling your garbage collector, this is common issue with scripts
and make sure you are on latest GIT build


On Thu, Aug 30, 2012 at 3:24 PM, Phillip Boles
<freeswitch-users at vocalspace.com> wrote:
> Sorry for the excessive logs. Here is my call to originate.
>
> var session = ManagedSession.OriginateHandleHangup(s, dialString, ts, (y)
> => {
> try {
> Log.WriteLine(LogLevel.Info, "Hanging UP: "+ y.GetUuid());
> y.SetAutoHangup(true);
> y.destroy();
>
> } catch( Exception ) {
> Log.WriteLine(LogLevel.Critical, "Exception While Trying to handup"); 
> }
> });
>
>
> My hangup callback is getting hit and I am destroying the session
>
> 2012-08-28 10:49:27.296108 [INFO] switch_cpp.cpp:1227 Handing UP:
> e315f2e8-1fa8-4fd9-849b-f687dad8aed5
>
> This is the only call on the system as it is a develpment machine and I see
> the call state being changed.
>
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:79
> sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682
> (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:407
> (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY
>
>
> If I call show channels after the above output it show there is a session
> sitting in CS_SOFT_EXEC corresponding to UUID
> e315f2e8-1fa8-4fd9-849b-f687dad8aed5.
> Is there something else I need to do to release the lock on this session to
> let the resources be reclaimed.
>
> Thanks!
>
> Phillip
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 grcamauer at gmail.com  Fri Aug 31 02:24:11 2012
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Thu, 30 Aug 2012 19:24:11 -0300
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <0D1C698866F66045A6201FD0F59CAC90014678ACA7@EX.frontier.local>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
	<CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
	<CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
	<CAA8_NKAnX97E0dV7tpakg+wO-TzWdKm5wrm0qNP1EqODAVRoNw@mail.gmail.com>
	<CAPNTnnpHzcUzJ8Jnkj6Nma_2pZVyVhfWDA4RjBtJkuNWEMOdvg@mail.gmail.com>
	<0D1C698866F66045A6201FD0F59CAC90014678ACA7@EX.frontier.local>
Message-ID: <CAPNTnnrbGTw8ywNgp8JJpAUApsptJB-YrcL3wee5SghH9MZHrQ@mail.gmail.com>

Performance problems as load increases which are not found on other distros
such as Debian.

On Thu, Aug 30, 2012 at 7:17 PM, Colin Mason <cmason at frontiernetworks.ca>wrote:

> What exactly is the problem?****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Guillermo
> Ruiz Camauer
> *Sent:* Thursday, August 30, 2012 6:13 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] RedHat 6.X Performance****
>
> ** **
>
> Is anyone having the same problems with RedHat or is this CentOS specific?
> ****
>
> ** **
>
> On Thu, Aug 30, 2012 at 6:23 PM, Gavin Henry <gavin.henry at gmail.com>
> wrote:****
>
> > Centos5 worked well for me, Centos6 worked for crap... I went to debian 6
> > (squeeze) and problems went a way... using the latest stable on the
> debian6
> > line installing from the netinstall (non-free variant as I have hardware
> > thats not supported under the main netinstall iso) and then installing
> > freeswitch via the instructions on the wiki and no problems so far****
>
> Same here. We're CentOS 5 here and Debian Squeezy for new boxes.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.surevoip.co.uk****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
>
>
> ****
>
> ** **
>
> --
> Guillermo Ruiz Camauer****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Guillermo Ruiz Camauer
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From mike.burlingame at me.com  Fri Aug 31 02:28:26 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Thu, 30 Aug 2012 15:28:26 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
References: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
Message-ID: <8D066FE8-DB24-4928-8508-317E23B858EA@me.com>

I would like to add an update to this and maybe someone has some other suggestions - It does look like that FS is disconnecting the call with cause code 31  [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on the Bye when I first looked at the log's

The basis of the issue still stands a 491 Results in the B-Leg being disconnected

On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame at me.com> wrote:

> We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
> 
> A-Leg
> Invite into Freeswitch
> 100 Trying back from FS to A-Leg
> 180 Ringing from FS to A-Leg
> 200 OK from FS to A-Leg at 15:08:14.638799
> Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515
> 100 Giving a try form A-Leg to FS at 15:08:14.749757
> 491 From A-Leg to FS at 15:08:14.780968
> ACK from FS to A-Leg at 15:08:14.781102
> ACK from FS to A-Leg at 15:08:14.797143
> BYE from A-LEG to FS 15:11:10.791963
   481 Call Does Not Exist back to A-LEG
> 
> B-Leg
> Invite from Freeswitch to B-Leg
> 100 Giving a try from B-Leg
> 180 Ringing from B-Leg
> 200 OK  from B-Leg at 15:08:14.635670
> ACK from FS to B-Leg  at 15:08:14.637044
> Invite from B-Leg to FS at 15:08:14.748623
> 100 Trying from FS to B-Leg at 15:08:14.748954
> 491 from FS to B-Leg at 15:08:14.782169
> ACK from B-Leg to FS at 15:08:14.782372
> BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31

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From cmason at frontiernetworks.ca  Fri Aug 31 02:31:10 2012
From: cmason at frontiernetworks.ca (Colin Mason)
Date: Thu, 30 Aug 2012 18:31:10 -0400
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAPNTnnrbGTw8ywNgp8JJpAUApsptJB-YrcL3wee5SghH9MZHrQ@mail.gmail.com>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
	<CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
	<CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
	<CAA8_NKAnX97E0dV7tpakg+wO-TzWdKm5wrm0qNP1EqODAVRoNw@mail.gmail.com>
	<CAPNTnnpHzcUzJ8Jnkj6Nma_2pZVyVhfWDA4RjBtJkuNWEMOdvg@mail.gmail.com>
	<0D1C698866F66045A6201FD0F59CAC90014678ACA7@EX.frontier.local>
	<CAPNTnnrbGTw8ywNgp8JJpAUApsptJB-YrcL3wee5SghH9MZHrQ@mail.gmail.com>
Message-ID: <0D1C698866F66045A6201FD0F59CAC90014678ACA9@EX.frontier.local>

Never had any performance issues on CentOS 6 under load. Can you be more specific?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Guillermo Ruiz Camauer
Sent: Thursday, August 30, 2012 6:24 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] RedHat 6.X Performance

Performance problems as load increases which are not found on other distros such as Debian.
On Thu, Aug 30, 2012 at 7:17 PM, Colin Mason <cmason at frontiernetworks.ca<mailto:cmason at frontiernetworks.ca>> wrote:
What exactly is 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 Guillermo Ruiz Camauer
Sent: Thursday, August 30, 2012 6:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] RedHat 6.X Performance

Is anyone having the same problems with RedHat or is this CentOS specific?

On Thu, Aug 30, 2012 at 6:23 PM, Gavin Henry <gavin.henry at gmail.com<mailto:gavin.henry at gmail.com>> wrote:
> Centos5 worked well for me, Centos6 worked for crap... I went to debian 6
> (squeeze) and problems went a way... using the latest stable on the debian6
> line installing from the netinstall (non-free variant as I have hardware
> thats not supported under the main netinstall iso) and then installing
> freeswitch via the instructions on the wiki and no problems so far
Same here. We're CentOS 5 here and Debian Squeezy for new boxes.

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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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



--
Guillermo Ruiz Camauer

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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



--
Guillermo Ruiz Camauer
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From egk at edjik.com  Fri Aug 31 02:59:45 2012
From: egk at edjik.com (Ed Kochanowski)
Date: Thu, 30 Aug 2012 15:59:45 -0700
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <0D1C698866F66045A6201FD0F59CAC90014678ACA9@EX.frontier.local>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
	<CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
	<CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
	<CAA8_NKAnX97E0dV7tpakg+wO-TzWdKm5wrm0qNP1EqODAVRoNw@mail.gmail.com>
	<CAPNTnnpHzcUzJ8Jnkj6Nma_2pZVyVhfWDA4RjBtJkuNWEMOdvg@mail.gmail.com>
	<0D1C698866F66045A6201FD0F59CAC90014678ACA7@EX.frontier.local>
	<CAPNTnnrbGTw8ywNgp8JJpAUApsptJB-YrcL3wee5SghH9MZHrQ@mail.gmail.com>
	<0D1C698866F66045A6201FD0F59CAC90014678ACA9@EX.frontier.local>
Message-ID: <CAFj9YCNA8WbKLssKheMSfQwPqN03omz2v-1BEvszj2dtAgdfuQ@mail.gmail.com>

We haven't had any issues either with Centos 6.3 kernel
2.6.32-279.el6.x86_64
Do you have a specific benchmark?

On Thu, Aug 30, 2012 at 3:31 PM, Colin Mason <cmason at frontiernetworks.ca>wrote:

> Never had any performance issues on CentOS 6 under load. Can you be more
> specific?****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Guillermo
> Ruiz Camauer
> *Sent:* Thursday, August 30, 2012 6:24 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] RedHat 6.X Performance****
>
> ** **
>
> Performance problems as load increases which are not found on other
> distros such as Debian.****
>
> On Thu, Aug 30, 2012 at 7:17 PM, Colin Mason <cmason at frontiernetworks.ca>
> wrote:****
>
> What exactly is the problem?****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Guillermo
> Ruiz Camauer
> *Sent:* Thursday, August 30, 2012 6:13 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] RedHat 6.X Performance****
>
>  ****
>
> Is anyone having the same problems with RedHat or is this CentOS specific?
> ****
>
>  ****
>
> On Thu, Aug 30, 2012 at 6:23 PM, Gavin Henry <gavin.henry at gmail.com>
> wrote:****
>
> > Centos5 worked well for me, Centos6 worked for crap... I went to debian 6
> > (squeeze) and problems went a way... using the latest stable on the
> debian6
> > line installing from the netinstall (non-free variant as I have hardware
> > thats not supported under the main netinstall iso) and then installing
> > freeswitch via the instructions on the wiki and no problems so far****
>
> Same here. We're CentOS 5 here and Debian Squeezy for new boxes.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.surevoip.co.uk****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
>
>
> ****
>
>  ****
>
> --
> Guillermo Ruiz Camauer****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
>
>
> ****
>
> ** **
>
> --
> Guillermo Ruiz Camauer****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 Aug 31 04:22:10 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 30 Aug 2012 19:22:10 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging in
 CS_SOFT_EXEC state
In-Reply-To: <20120830221839.7f6cf44b@mail.tritonwest.net>
References: <FB1F0348-B48D-44FA-821D-1473E4EEDCE1@vocalspace.com>
	<20120830221839.7f6cf44b@mail.tritonwest.net>
Message-ID: <CAKbxfG8fRdhkAq_CQMpi+jXcap7vFt5z31sv=rSVQpa0Ko5Fhg@mail.gmail.com>

What do you mean? Everyone has access to the wiki.  I would think that
after all these years you would have an account by now.
On Aug 30, 2012 5:19 PM, "Dave R. Kompel" <drk at drkngs.net> wrote:

> **
> Argh, I wish someone with access to the wiki would fix this.......
>
> The old methods for making outbound legs from mod_managed, are just that,
> old. Where are you calling this from? If you want to originate a call from
> managed code the only way you won't run into some problem with
> cross_app_domain stuff, is to use Api.Execute("originate",...), and if you
> want to bridge a call then use Session.Execute("Bridge...").
>
> If you stick to this, you should have no problem. If you need a managed
> session handle when you do an originate, then have the Originate API call a
> managed dialplan app, and it can do the rest.
>
> --Dave
>
>  ------------------------------
> *From:* Phillip Boles [mailto:freeswitch-users at vocalspace.com]
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thu, 30 Aug 2012 11:20:06 -0700
> *Subject:* [Freeswitch-users] Problem with originated calls hanging in
> CS_SOFT_EXEC state
>
> 	
> 	var session =  ManagedSession.OriginateHandleHangup(s, dialString, ts, (y) => {
> 		try {
> 			Log.WriteLine(LogLevel.Info <http://loglevel.info/>, "Hanging UP: "+ y.GetUuid());
> 			y.SetAutoHangup(true);
> 			y.destroy();
>
> 		} catch( Exception ) {
> 			Log.WriteLine(LogLevel.Critical, "Exception While Trying to handup");	
> 		}
> 	});
> 			
>
> 	
> 2012-08-28 10:49:12.156172 [INFO] switch_cpp.cpp:1227 Dialing
> 2012-08-28 10:49:12.156172 [DEBUG] switch_ivr_originate.c:1947 Parsing global variables
> 2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [ignore_early_media]=[true]
> 2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [origination_caller_id_number]=[2223334444]
> 2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [origination_caller_id_name]=[Vocalspace]
> 2012-08-28 10:49:12.156172 [NOTICE] switch_channel.c:926 New Channel sofia/external/XXXXXXXXXXX [e315f2e8-1fa8-4fd9-849b-f687dad8aed5]
> 2012-08-28 10:49:12.156172 [DEBUG] mod_sofia.c:4709 (sofia/external/XXXXXXXXXXX) State Change CS_NEW -> CS_INIT
> 2012-08-28 10:49:12.156172 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:12.156172 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_INIT
> 2012-08-28 10:49:12.156172 [DEBUG] switch_core_state_machine.c:421 (sofia/external/XXXXXXXXXXX) State INIT
> 2012-08-28 10:49:12.156172 [DEBUG] mod_sofia.c:85 sofia/external/XXXXXXXXXXX SOFIA INIT
> 2012-08-28 10:49:12.216161 [DEBUG] switch_nat.c:511 mapped public port 19628 protocol UDP to localport 19628
> 2012-08-28 10:49:12.256159 [DEBUG] switch_nat.c:511 mapped public port 19629 protocol UDP to localport 19629
> 2012-08-28 10:49:12.256159 [DEBUG] mod_sofia.c:129 (sofia/external/XXXXXXXXXXX) State Change CS_INIT -> CS_ROUTING
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:421 (sofia/external/XXXXXXXXXXX) State INIT going to sleep
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_ROUTING
> 2012-08-28 10:49:12.256159 [DEBUG] switch_channel.c:1887 (sofia/external/XXXXXXXXXXX) Callstate Change DOWN -> RINGING
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:430 (sofia/external/XXXXXXXXXXX) State ROUTING
> 2012-08-28 10:49:12.256159 [DEBUG] mod_sofia.c:152 sofia/external/XXXXXXXXXXX SOFIA ROUTING
> 2012-08-28 10:49:12.256159 [DEBUG] switch_ivr_originate.c:67 (sofia/external/XXXXXXXXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:430 (sofia/external/XXXXXXXXXXX) State ROUTING going to sleep
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_CONSUME_MEDIA
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:449 (sofia/external/XXXXXXXXXXX) State CONSUME_MEDIA
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:449 (sofia/external/XXXXXXXXXXX) State CONSUME_MEDIA going to sleep
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:12.256159 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [calling][0]
> 2012-08-28 10:49:12.636131 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:12.636131 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:12.636131 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call"
> 2012-08-28 10:49:12.636131 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [proceeding][180]
> 2012-08-28 10:49:12.636131 [NOTICE] sofia.c:5769 Ring-Ready sofia/external/XXXXXXXXXXX!
> 2012-08-28 10:49:14.655990 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:14.655990 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:14.655990 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call"
> 2012-08-28 10:49:14.655990 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [proceeding][183]
> 2012-08-28 10:49:14.655990 [DEBUG] sofia.c:5688 Remote SDP:
> v=0
> o=FreeSWITCH 1346150407 1346150408 IN IP4 8.19.97.6
> s=FreeSWITCH
> c=IN IP4 8.19.97.6
> t=0 0
> m=audio 18534 RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:4911 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:2995 Set Codec sofia/external/XXXXXXXXXXX PCMU/8000 20 ms 160 samples 64000 bits
> 2012-08-28 10:49:14.655990 [DEBUG] switch_core_codec.c:111 sofia/external/XXXXXXXXXXX Original read codec set to PCMU:0
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:5025 Set 2833 dtmf send payload to 101
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3244 AUDIO RTP [sofia/external/XXXXXXXXXXX] 10.10.80.33 port 19628 -> 8.19.97.6 port 18534 codec: 0 ms: 20
> 2012-08-28 10:49:14.655990 [DEBUG] switch_rtp.c:1676 Starting timer [soft] 160 bytes per 20ms
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3508 Set 2833 dtmf send payload to 101
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3514 Set 2833 dtmf receive payload to 101
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3541 sofia/external/XXXXXXXXXXX Set rtp dtmf delay to 40
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3547 Set comfort noise payload to 13
> 2012-08-28 10:49:14.655990 [NOTICE] sofia_glue.c:4052 Pre-Answer sofia/external/XXXXXXXXXXX!
> 2012-08-28 10:49:14.655990 [DEBUG] switch_channel.c:2986 (sofia/external/XXXXXXXXXXX) Callstate Change RINGING -> EARLY
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:22.136467 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call"
> 2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [completing][200]
> 2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5685 Duplicate SDP
> v=0
> o=FreeSWITCH 1346150407 1346150408 IN IP4 8.19.97.6
> s=FreeSWITCH
> c=IN IP4 8.19.97.6
> t=0 0
> m=audio 18534 RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [ready][200]
> 2012-08-28 10:49:22.136467 [DEBUG] switch_channel.c:3245 (sofia/external/XXXXXXXXXXX) Callstate Change EARLY -> ACTIVE
> 2012-08-28 10:49:22.136467 [NOTICE] sofia.c:6352 Channel [sofia/external/XXXXXXXXXXX] has been answered
> 2012-08-28 10:49:22.136467 [DEBUG] switch_ivr_originate.c:3330 Originate Resulted in Success: [sofia/external/XXXXXXXXXXX]
> 2012-08-28 10:49:22.136467 [DEBUG] switch_cpp.cpp:1083 (sofia/external/XXXXXXXXXXX) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_SOFT_EXECUTE
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:443 (sofia/external/XXXXXXXXXXX) State SOFT_EXECUTE
> 2012-08-28 10:49:22.136467 [DEBUG] mod_sofia.c:592 SOFIA SOFT_EXECUTE
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:261 sofia/external/XXXXXXXXXXX Standard SOFT_EXECUTE
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:443 (sofia/external/XXXXXXXXXXX) State SOFT_EXECUTE going to sleep
> 2012-08-28 10:49:22.136467 [DEBUG] switch_cpp.cpp:1172 CoreSession::seHangupHook, hangup_func: (nil)
> 2012-08-28 10:49:22.136467 [INFO] switch_cpp.cpp:1227 Connected
> 2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Session Ready
> 2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Call Initialized
> 2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Trying To Run
> 2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Running
> 2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Checking for file /svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw
> 2012-08-28 10:49:23.160407 [ERR] mod_sndfile.c:197 Error Opening File [/svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw] [System error : No such file or directory.]
> 2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Transcoding xfile /svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw
> 2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Sending MQ Message
> 2012-08-28 10:49:23.296392 [NOTICE] switch_cpp.cpp:1227 Creating a channel...
> 2012-08-28 10:49:23.296392 [NOTICE] switch_cpp.cpp:1227 Defining Exchange FS...
> 2012-08-28 10:49:23.316389 [NOTICE] switch_cpp.cpp:1227 Binding Exchange and Queue..
> 2012-08-28 10:49:23.336389 [NOTICE] switch_cpp.cpp:1227 Sending message..
> 2012-08-28 10:49:23.356388 [NOTICE] switch_cpp.cpp:1227 shout://www.product.com/files/media/e53bc5bf-4e00-ab8d-4e4d-b201699628d1.mp3
> 2012-08-28 10:49:25.916204 [DEBUG] mod_shout.c:472 Read Thread Done
> 2012-08-28 10:49:26.356173 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
> 2012-08-28 10:49:26.596156 [DEBUG] switch_rtp.c:3252 Correct ip/port confirmed.
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:27.296108 [DEBUG] switch_channel.c:2849 (sofia/external/XXXXXXXXXXX) Callstate Change ACTIVE -> HANGUP
> 2012-08-28 10:49:27.296108 [NOTICE] sofia.c:694 Hangup sofia/external/XXXXXXXXXXX [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
> 2012-08-28 10:49:27.296108 [DEBUG] switch_channel.c:2872 Send signal sofia/external/XXXXXXXXXXX [KILL]
> 2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1227 AppFunction is in hangupCallback.
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_HANGUP
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:622 (sofia/external/XXXXXXXXXXX) State HANGUP
> 2012-08-28 10:49:27.296108 [DEBUG] mod_sofia.c:473 Channel sofia/external/XXXXXXXXXXX hanging up, cause: NORMAL_CLEARING
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:47 sofia/external/XXXXXXXXXXX Standard HANGUP, cause: NORMAL_CLEARING
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:622 (sofia/external/XXXXXXXXXXX) State HANGUP going to sleep
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:413 (sofia/external/XXXXXXXXXXX) State Change CS_HANGUP -> CS_REPORTING
> 2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1227 AppFunction is in hangupCallback.
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_REPORTING
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682 (sofia/external/XXXXXXXXXXX) State REPORTING
> 2012-08-28 10:49:27.296108 [INFO] switch_cpp.cpp:1227 Handing UP: e315f2e8-1fa8-4fd9-849b-f687dad8aed5
> 2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1007 sofia/external/XXXXXXXXXXX destroy/unlink session from object
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:79 sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682 (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:407 (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1424 Session 4 (sofia/external/XXXXXXXXXXX) Locked, Waiting on external entities
> 2012-08-28 10:49:27.316110 [DEBUG] switch_ivr_play_say.c:1682 done playing file shout://www.product.com/files/media/e53bc5bf-4e00-ab8d-4e4d-b201699628d1.mp3
> freeswitch at fs03.int.colo> show channels
>
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
> e315f2e8-1fa8-4fd9-849b-f687dad8aed5,outbound,2012-08-28 10:49:12,1346168952,sofia/external/XXXXXXXXXXX,CS_SOFT_EXECUTE,Vocalspace,2223334444,,XXXXXXXXXXX,,,,default,PCMU,8000,64000,PCMU,8000,64000,,fs03.int.colo,,,ACTIVE,Outbound Call,XXXXXXXXXXX,,,,
>
> 1 total.
>
> 2012-08-28 10:58:10.256585 [DEBUG] switch_nat.c:511 mapped public port 5060 protocol TCP to localport 5060
> 2012-08-28 10:58:10.296583 [DEBUG] switch_nat.c:511 mapped public port 5080 protocol UDP to localport 5080
> 2012-08-28 10:58:10.336580 [DEBUG] switch_nat.c:511 mapped public port 5080 protocol TCP to localport 5080
> 2012-08-28 10:58:10.396576 [DEBUG] switch_nat.c:511 mapped public port 19628 protocol UDP to localport 19628
> 2012-08-28 10:58:10.436573 [DEBUG] switch_nat.c:511 mapped public port 19629 protocol UDP to localport 19629
> 2012-08-28 10:58:10.476570 [DEBUG] switch_nat.c:511 mapped public port 22013 protocol UDP to localport 22013
> 2012-08-28 10:58:10.536566 [DEBUG] switch_nat.c:511 mapped public port 22191 protocol UDP to localport 22191
> 2012-08-28 10:58:10.576563 [DEBUG] switch_nat.c:511 mapped public port 25326 protocol UDP to localport 25326
> 2012-08-28 10:58:10.636559 [DEBUG] switch_nat.c:511 mapped public port 25327 protocol UDP to localport 25327
> Loading /usr/lib64/freeswitch/mod/managed/Enyim.Caching.dll from domain Enyim.Caching.dll_63
> 2012-08-28 11:00:02.097780 [DEBUG] switch_cpp.cpp:1227 Assembly Enyim.Caching, Version=1.2.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/Jint.dll from domain Jint.dll_64
> 2012-08-28 11:00:02.316760 [DEBUG] switch_cpp.cpp:1227 Assembly Jint, Version=0.9.2.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/log4net.dll from domain log4net.dll_65
> 2012-08-28 11:00:02.536745 [DEBUG] switch_cpp.cpp:1227 Assembly log4net, Version=1.2.10.0, Culture=neutral, PublicKeyToken=1b44e1d426115821 doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/Antlr3.Runtime.dll from domain Antlr3.Runtime.dll_66
> 2012-08-28 11:00:02.756720 [DEBUG] switch_cpp.cpp:1227 Assembly Antlr3.Runtime, Version=3.3.1.7705, Culture=neutral, PublicKeyToken=eb42632606e9261f doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/RabbitMQ.Client.dll from domain RabbitMQ.Client.dll_67
> 2012-08-28 11:00:02.996708 [DEBUG] switch_cpp.cpp:1227 Assembly RabbitMQ.Client, Version=0.0.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/ManagedEsl.dll from domain ManagedEsl.dll_68
> 2012-08-28 11:00:03.216689 [DEBUG] switch_cpp.cpp:1227 Assembly ManagedEsl, Version=1.0.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/FreeSWITCH.Managed.dll from domain FreeSWITCH.Managed.dll_69
> 2012-08-28 11:00:03.436678 [DEBUG] switch_cpp.cpp:1227 Assembly FreeSWITCH.Managed, Version=1.0.5.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/FreeSwitch.EventSocket.dll from domain FreeSwitch.EventSocket.dll_70
> 2012-08-28 11:00:03.657671 [DEBUG] switch_cpp.cpp:1227 Assembly FreeSwitch.EventSocket, Version=1.0.0.0, Culture=neutral, PublicKeyToken=94d8a22cdb4b7713 doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/Newtonsoft.Json.dll from domain Newtonsoft.Json.dll_71
> 2012-08-28 11:00:03.876655 [DEBUG] switch_cpp.cpp:1227 Assembly Newtonsoft.Json, Version=4.5.0.0, Culture=neutral, PublicKeyToken=30ad4fe6b2a6aeed doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/Freeswitch.UUID.dll from domain Freeswitch.UUID.dll_72
> 2012-08-28 11:00:04.096628 [DEBUG] switch_cpp.cpp:1227 Assembly Freeswitch.UUID, Version=1.0.0.0, Culture=neutral, PublicKeyToken=f758751257b2cdaf doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/Product.dll from domain Product.dll_73
> freeswitch at fs03.int.colo> show channels
>
>
> 0 total.
>
>
> 	
> 	var session =  ManagedSession.OriginateHandleHangup(s, dialString, ts, (y) => {
> 		try {
> 			Log.WriteLine(LogLevel.Info <http://loglevel.info/>, "Hanging UP: "+ y.GetUuid());
> 			y.SetAutoHangup(true);
> 			y.destroy();
>
> 		} catch( Exception ) {
> 			Log.WriteLine(LogLevel.Critical, "Exception While Trying to handup");	
> 		}
> 	});
> 			
>
> 	
> 2012-08-28 10:49:12.156172 [INFO] switch_cpp.cpp:1227 Dialing
> 2012-08-28 10:49:12.156172 [DEBUG] switch_ivr_originate.c:1947 Parsing global variables
> 2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [ignore_early_media]=[true]
> 2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [origination_caller_id_number]=[2223334444]
> 2012-08-28 10:49:12.156172 [DEBUG] switch_event.c:1527 Parsing variable [origination_caller_id_name]=[Vocalspace]
> 2012-08-28 10:49:12.156172 [NOTICE] switch_channel.c:926 New Channel sofia/external/XXXXXXXXXXX [e315f2e8-1fa8-4fd9-849b-f687dad8aed5]
> 2012-08-28 10:49:12.156172 [DEBUG] mod_sofia.c:4709 (sofia/external/XXXXXXXXXXX) State Change CS_NEW -> CS_INIT
> 2012-08-28 10:49:12.156172 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:12.156172 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_INIT
> 2012-08-28 10:49:12.156172 [DEBUG] switch_core_state_machine.c:421 (sofia/external/XXXXXXXXXXX) State INIT
> 2012-08-28 10:49:12.156172 [DEBUG] mod_sofia.c:85 sofia/external/XXXXXXXXXXX SOFIA INIT
> 2012-08-28 10:49:12.216161 [DEBUG] switch_nat.c:511 mapped public port 19628 protocol UDP to localport 19628
> 2012-08-28 10:49:12.256159 [DEBUG] switch_nat.c:511 mapped public port 19629 protocol UDP to localport 19629
> 2012-08-28 10:49:12.256159 [DEBUG] mod_sofia.c:129 (sofia/external/XXXXXXXXXXX) State Change CS_INIT -> CS_ROUTING
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:421 (sofia/external/XXXXXXXXXXX) State INIT going to sleep
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_ROUTING
> 2012-08-28 10:49:12.256159 [DEBUG] switch_channel.c:1887 (sofia/external/XXXXXXXXXXX) Callstate Change DOWN -> RINGING
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:430 (sofia/external/XXXXXXXXXXX) State ROUTING
> 2012-08-28 10:49:12.256159 [DEBUG] mod_sofia.c:152 sofia/external/XXXXXXXXXXX SOFIA ROUTING
> 2012-08-28 10:49:12.256159 [DEBUG] switch_ivr_originate.c:67 (sofia/external/XXXXXXXXXXX) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:430 (sofia/external/XXXXXXXXXXX) State ROUTING going to sleep
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_CONSUME_MEDIA
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:449 (sofia/external/XXXXXXXXXXX) State CONSUME_MEDIA
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_state_machine.c:449 (sofia/external/XXXXXXXXXXX) State CONSUME_MEDIA going to sleep
> 2012-08-28 10:49:12.256159 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:12.256159 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [calling][0]
> 2012-08-28 10:49:12.636131 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:12.636131 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:12.636131 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call"
> 2012-08-28 10:49:12.636131 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [proceeding][180]
> 2012-08-28 10:49:12.636131 [NOTICE] sofia.c:5769 Ring-Ready sofia/external/XXXXXXXXXXX!
> 2012-08-28 10:49:14.655990 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:14.655990 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:14.655990 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call"
> 2012-08-28 10:49:14.655990 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [proceeding][183]
> 2012-08-28 10:49:14.655990 [DEBUG] sofia.c:5688 Remote SDP:
> v=0
> o=FreeSWITCH 1346150407 1346150408 IN IP4 8.19.97.6
> s=FreeSWITCH
> c=IN IP4 8.19.97.6
> t=0 0
> m=audio 18534 RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:4911 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:2995 Set Codec sofia/external/XXXXXXXXXXX PCMU/8000 20 ms 160 samples 64000 bits
> 2012-08-28 10:49:14.655990 [DEBUG] switch_core_codec.c:111 sofia/external/XXXXXXXXXXX Original read codec set to PCMU:0
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:5025 Set 2833 dtmf send payload to 101
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3244 AUDIO RTP [sofia/external/XXXXXXXXXXX] 10.10.80.33 port 19628 -> 8.19.97.6 port 18534 codec: 0 ms: 20
> 2012-08-28 10:49:14.655990 [DEBUG] switch_rtp.c:1676 Starting timer [soft] 160 bytes per 20ms
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3508 Set 2833 dtmf send payload to 101
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3514 Set 2833 dtmf receive payload to 101
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3541 sofia/external/XXXXXXXXXXX Set rtp dtmf delay to 40
> 2012-08-28 10:49:14.655990 [DEBUG] sofia_glue.c:3547 Set comfort noise payload to 13
> 2012-08-28 10:49:14.655990 [NOTICE] sofia_glue.c:4052 Pre-Answer sofia/external/XXXXXXXXXXX!
> 2012-08-28 10:49:14.655990 [DEBUG] switch_channel.c:2986 (sofia/external/XXXXXXXXXXX) Callstate Change RINGING -> EARLY
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:22.136467 [INFO] sofia.c:890 sofia/external/XXXXXXXXXXX Update Callee ID to "Outbound Call"
> 2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [completing][200]
> 2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5685 Duplicate SDP
> v=0
> o=FreeSWITCH 1346150407 1346150408 IN IP4 8.19.97.6
> s=FreeSWITCH
> c=IN IP4 8.19.97.6
> t=0 0
> m=audio 18534 RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:22.136467 [DEBUG] sofia.c:5677 Channel sofia/external/XXXXXXXXXXX entering state [ready][200]
> 2012-08-28 10:49:22.136467 [DEBUG] switch_channel.c:3245 (sofia/external/XXXXXXXXXXX) Callstate Change EARLY -> ACTIVE
> 2012-08-28 10:49:22.136467 [NOTICE] sofia.c:6352 Channel [sofia/external/XXXXXXXXXXX] has been answered
> 2012-08-28 10:49:22.136467 [DEBUG] switch_ivr_originate.c:3330 Originate Resulted in Success: [sofia/external/XXXXXXXXXXX]
> 2012-08-28 10:49:22.136467 [DEBUG] switch_cpp.cpp:1083 (sofia/external/XXXXXXXXXXX) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_SOFT_EXECUTE
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:443 (sofia/external/XXXXXXXXXXX) State SOFT_EXECUTE
> 2012-08-28 10:49:22.136467 [DEBUG] mod_sofia.c:592 SOFIA SOFT_EXECUTE
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:261 sofia/external/XXXXXXXXXXX Standard SOFT_EXECUTE
> 2012-08-28 10:49:22.136467 [DEBUG] switch_core_state_machine.c:443 (sofia/external/XXXXXXXXXXX) State SOFT_EXECUTE going to sleep
> 2012-08-28 10:49:22.136467 [DEBUG] switch_cpp.cpp:1172 CoreSession::seHangupHook, hangup_func: (nil)
> 2012-08-28 10:49:22.136467 [INFO] switch_cpp.cpp:1227 Connected
> 2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Session Ready
> 2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Call Initialized
> 2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Trying To Run
> 2012-08-28 10:49:23.136397 [INFO] switch_cpp.cpp:1227 Running
> 2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Checking for file /svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw
> 2012-08-28 10:49:23.160407 [ERR] mod_sndfile.c:197 Error Opening File [/svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw] [System error : No such file or directory.]
> 2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Transcoding xfile /svc/prod/product/2167/phone/2167.a74e3d86b1f93f0a1f2fc9ad4f6ef2f4.ulaw.ulaw
> 2012-08-28 10:49:23.160407 [NOTICE] switch_cpp.cpp:1227 Sending MQ Message
> 2012-08-28 10:49:23.296392 [NOTICE] switch_cpp.cpp:1227 Creating a channel...
> 2012-08-28 10:49:23.296392 [NOTICE] switch_cpp.cpp:1227 Defining Exchange FS...
> 2012-08-28 10:49:23.316389 [NOTICE] switch_cpp.cpp:1227 Binding Exchange and Queue..
> 2012-08-28 10:49:23.336389 [NOTICE] switch_cpp.cpp:1227 Sending message..
> 2012-08-28 10:49:23.356388 [NOTICE] switch_cpp.cpp:1227 shout://www.product.com/files/media/e53bc5bf-4e00-ab8d-4e4d-b201699628d1.mp3
> 2012-08-28 10:49:25.916204 [DEBUG] mod_shout.c:472 Read Thread Done
> 2012-08-28 10:49:26.356173 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
> 2012-08-28 10:49:26.596156 [DEBUG] switch_rtp.c:3252 Correct ip/port confirmed.
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:919 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:27.296108 [DEBUG] switch_channel.c:2849 (sofia/external/XXXXXXXXXXX) Callstate Change ACTIVE -> HANGUP
> 2012-08-28 10:49:27.296108 [NOTICE] sofia.c:694 Hangup sofia/external/XXXXXXXXXXX [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
> 2012-08-28 10:49:27.296108 [DEBUG] switch_channel.c:2872 Send signal sofia/external/XXXXXXXXXXX [KILL]
> 2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1227 AppFunction is in hangupCallback.
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_HANGUP
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:622 (sofia/external/XXXXXXXXXXX) State HANGUP
> 2012-08-28 10:49:27.296108 [DEBUG] mod_sofia.c:473 Channel sofia/external/XXXXXXXXXXX hanging up, cause: NORMAL_CLEARING
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:47 sofia/external/XXXXXXXXXXX Standard HANGUP, cause: NORMAL_CLEARING
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:622 (sofia/external/XXXXXXXXXXX) State HANGUP going to sleep
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:413 (sofia/external/XXXXXXXXXXX) State Change CS_HANGUP -> CS_REPORTING
> 2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1227 AppFunction is in hangupCallback.
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:382 (sofia/external/XXXXXXXXXXX) Running State Change CS_REPORTING
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682 (sofia/external/XXXXXXXXXXX) State REPORTING
> 2012-08-28 10:49:27.296108 [INFO] switch_cpp.cpp:1227 Handing UP: e315f2e8-1fa8-4fd9-849b-f687dad8aed5
> 2012-08-28 10:49:27.296108 [DEBUG] switch_cpp.cpp:1007 sofia/external/XXXXXXXXXXX destroy/unlink session from object
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:79 sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682 (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:407 (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXXX [BREAK]
> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_session.c:1424 Session 4 (sofia/external/XXXXXXXXXXX) Locked, Waiting on external entities
> 2012-08-28 10:49:27.316110 [DEBUG] switch_ivr_play_say.c:1682 done playing file shout://www.product.com/files/media/e53bc5bf-4e00-ab8d-4e4d-b201699628d1.mp3
> freeswitch at fs03.int.colo> show channels
>
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
> e315f2e8-1fa8-4fd9-849b-f687dad8aed5,outbound,2012-08-28 10:49:12,1346168952,sofia/external/XXXXXXXXXXX,CS_SOFT_EXECUTE,Vocalspace,2223334444,,XXXXXXXXXXX,,,,default,PCMU,8000,64000,PCMU,8000,64000,,fs03.int.colo,,,ACTIVE,Outbound Call,XXXXXXXXXXX,,,,
>
> 1 total.
>
> 2012-08-28 10:58:10.256585 [DEBUG] switch_nat.c:511 mapped public port 5060 protocol TCP to localport 5060
> 2012-08-28 10:58:10.296583 [DEBUG] switch_nat.c:511 mapped public port 5080 protocol UDP to localport 5080
> 2012-08-28 10:58:10.336580 [DEBUG] switch_nat.c:511 mapped public port 5080 protocol TCP to localport 5080
> 2012-08-28 10:58:10.396576 [DEBUG] switch_nat.c:511 mapped public port 19628 protocol UDP to localport 19628
> 2012-08-28 10:58:10.436573 [DEBUG] switch_nat.c:511 mapped public port 19629 protocol UDP to localport 19629
> 2012-08-28 10:58:10.476570 [DEBUG] switch_nat.c:511 mapped public port 22013 protocol UDP to localport 22013
> 2012-08-28 10:58:10.536566 [DEBUG] switch_nat.c:511 mapped public port 22191 protocol UDP to localport 22191
> 2012-08-28 10:58:10.576563 [DEBUG] switch_nat.c:511 mapped public port 25326 protocol UDP to localport 25326
> 2012-08-28 10:58:10.636559 [DEBUG] switch_nat.c:511 mapped public port 25327 protocol UDP to localport 25327
> Loading /usr/lib64/freeswitch/mod/managed/Enyim.Caching.dll from domain Enyim.Caching.dll_63
> 2012-08-28 11:00:02.097780 [DEBUG] switch_cpp.cpp:1227 Assembly Enyim.Caching, Version=1.2.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/Jint.dll from domain Jint.dll_64
> 2012-08-28 11:00:02.316760 [DEBUG] switch_cpp.cpp:1227 Assembly Jint, Version=0.9.2.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/log4net.dll from domain log4net.dll_65
> 2012-08-28 11:00:02.536745 [DEBUG] switch_cpp.cpp:1227 Assembly log4net, Version=1.2.10.0, Culture=neutral, PublicKeyToken=1b44e1d426115821 doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/Antlr3.Runtime.dll from domain Antlr3.Runtime.dll_66
> 2012-08-28 11:00:02.756720 [DEBUG] switch_cpp.cpp:1227 Assembly Antlr3.Runtime, Version=3.3.1.7705, Culture=neutral, PublicKeyToken=eb42632606e9261f doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/RabbitMQ.Client.dll from domain RabbitMQ.Client.dll_67
> 2012-08-28 11:00:02.996708 [DEBUG] switch_cpp.cpp:1227 Assembly RabbitMQ.Client, Version=0.0.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/ManagedEsl.dll from domain ManagedEsl.dll_68
> 2012-08-28 11:00:03.216689 [DEBUG] switch_cpp.cpp:1227 Assembly ManagedEsl, Version=1.0.0.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/FreeSWITCH.Managed.dll from domain FreeSWITCH.Managed.dll_69
> 2012-08-28 11:00:03.436678 [DEBUG] switch_cpp.cpp:1227 Assembly FreeSWITCH.Managed, Version=1.0.5.0, Culture=neutral, PublicKeyToken=null doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/FreeSwitch.EventSocket.dll from domain FreeSwitch.EventSocket.dll_70
> 2012-08-28 11:00:03.657671 [DEBUG] switch_cpp.cpp:1227 Assembly FreeSwitch.EventSocket, Version=1.0.0.0, Culture=neutral, PublicKeyToken=94d8a22cdb4b7713 doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/Newtonsoft.Json.dll from domain Newtonsoft.Json.dll_71
> 2012-08-28 11:00:03.876655 [DEBUG] switch_cpp.cpp:1227 Assembly Newtonsoft.Json, Version=4.5.0.0, Culture=neutral, PublicKeyToken=30ad4fe6b2a6aeed doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/Freeswitch.UUID.dll from domain Freeswitch.UUID.dll_72
> 2012-08-28 11:00:04.096628 [DEBUG] switch_cpp.cpp:1227 Assembly Freeswitch.UUID, Version=1.0.0.0, Culture=neutral, PublicKeyToken=f758751257b2cdaf doesn't reference FreeSWITCH.Managed, not loading.
> Loading /usr/lib64/freeswitch/mod/managed/Product.dll from domain Product.dll_73
> freeswitch at fs03.int.colo> show channels
>
>
> 0 total.
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 dujinfang at gmail.com  Fri Aug 31 04:27:23 2012
From: dujinfang at gmail.com (dujinfang)
Date: Fri, 31 Aug 2012 08:27:23 +0800
Subject: [Freeswitch-users] Best way to stream record session?
In-Reply-To: <CAKzWOxXDTUGWv4OitdRozX_DKWDmptiVwaBH8D+Bz2XDfN-umw@mail.gmail.com>
References: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FC202C@BLUPRD0711MB413.namprd07.prod.outlook.com>
	<000001cd86a8$75efcd10$61cf6730$@207me.com>
	<CAKzWOxXDTUGWv4OitdRozX_DKWDmptiVwaBH8D+Bz2XDfN-umw@mail.gmail.com>
Message-ID: <18755CF1-687D-4BCA-AE57-DE2FB72B2ED0@gmail.com>

maybe u could also try the Udp endpoint in Sofia

???? iPhone

? 2012-8-31???12:19?Michael Collins <msc at freeswitch.org> ???

> 
> 
> On Thu, Aug 30, 2012 at 5:10 AM, Stephen Dame <sdame at 207me.com> wrote:
> Mod_shout  configured  to an icecast server works well for this.   The wiki
> has examples.
> +1 - this works very well in my experience.
> -MC
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 Aug 31 04:29:51 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 30 Aug 2012 19:29:51 -0500
Subject: [Freeswitch-users] endpoint /user and sofia_contact change?
In-Reply-To: <CALNZuwmmm_3yo2tdYT_58M7Qgbz7QiRzmf42kpCwta5psD4YFg@mail.gmail.com>
References: <CALNZuwnv86enuhdOm57zSwF7G-D9U4gFnUBZR8R+QWjC5Wu4zw@mail.gmail.com>
	<CAC-LwPMoGO2GCeMxC3Yw6-FV6wzwA6LvTyFKbmWt9O8s7Ua9pw@mail.gmail.com>
	<CALNZuwmmm_3yo2tdYT_58M7Qgbz7QiRzmf42kpCwta5psD4YFg@mail.gmail.com>
Message-ID: <CAKbxfG9QYWebJ0RkH21yYxD2sLjjhJ+-7PYNVLJ8aNe2zU6kFA@mail.gmail.com>

Reverse that, a bug was fixed that may now appear as a bug, add the special
profile name * in your sofia_contact call so it searches all profiles...

sofia_contact */user at domain.com
On Aug 29, 2012 10:36 AM, "Avi Marcus" <avi at avimarcus.net> wrote:

> freeswitch at default> sofia_contact 1000
> error/user_not_registered
>
> freeswitch at default> sofia_contact 1000 at sip.mydomain.com
> error/user_not_registered
>
> freeswitch at default> sofia_contact internal/1000 at sip.mydomain.com
> sofia/internal/sip:1000 at 192.168.255.103:5077,sofia/internal/
> sip:1000 at 192.168.255.103:5078
> ,sofia/internal/sip:1000 at 192.168.255.103:32998;ob
>
> The first 2 worked before. I suppose this is a bug.
>
> -Avi
>
>
> On Wed, Aug 29, 2012 at 6:23 PM, Vik Killa <vipkilla at gmail.com> wrote:
>
>> does 'sofia status profile <profile_name> reg' show them as registered?
>> try registering the handsets or flushing sofia registrations and
>> re-registering phone.
>>
>> On Wed, Aug 29, 2012 at 11:07 AM, Avi Marcus <avi at avimarcus.net> wrote:
>> > I think I had 1.2 release yesterday and I would bridge to user/1000 or
>> user
>> > 1000 at mydomain.com and/or do sofia_contact(1000) and all is fine.
>> > However, since building on master in GIT last night: FreeSWITCH Version
>> > 1.3.0+git~20120828T230559Z~f1d201f03b (1.3.0; git at commit f1d201f03b
>> on
>> > Tue, 28 Aug 2012 23:05:59 Z)
>> >
>> > They return not registered. I have to do sofia_contact internal/1000.
>> I'm
>> > not sure how to use user/ I replaced it with
>> > ${sofia_contact(internal/1000)}.
>> > Was this an intentional change?
>> > Is there some list of breaking changes somewhere?
>> >
>> > -Avi
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.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
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 Aug 31 05:07:17 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 18:07:17 -0700
Subject: [Freeswitch-users] cisco spa 500 configs?
Message-ID: <CAKzWOxWc68uMp0R+NCVATm6YZfxT6DEn7v9whwoyMKBDzXWkmg@mail.gmail.com>

Does anyone have a working config for a spa 5xx phone w/ s500 sidecar? I'd
like to get that and the SPA 300 phones better documented on the wiki.

Thanks!

-MC

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From anthony.minessale at gmail.com  Fri Aug 31 05:19:08 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 30 Aug 2012 20:19:08 -0500
Subject: [Freeswitch-users] uuid_broadcast muxing.
In-Reply-To: <503EF955.1080400@gmail.com>
References: <503E39C6.40306@gmail.com>
	<CAKbxfG9CsU4U+xEYmVBWrGdXDDdsSfuXQEwQW7HP5bJQtogLGA@mail.gmail.com>
	<503EF955.1080400@gmail.com>
Message-ID: <CAKbxfG92Qiw54Grik6mqEP4+GM6+WNN=a2TEuDx-a94+RDdd0A@mail.gmail.com>

nobody is actually hearing it discard the audio its only the audio to
the handle where you have the music paused.
A normal file handle waits for you to read, the local_stream one has a
buffer feeding it from a master stream, so when you don't read from
it, you lose the audio because the music is playing on without you
even when you pause.


On Thu, Aug 30, 2012 at 12:25 AM, Vbvbrj <vbvbrj at gmail.com> wrote:
> On 30.08.2012 02:21, Anthony Minessale wrote:
>> The error is harmless warning to remind you the local_stream is being ignored.
>>
>>
>> On Wed, Aug 29, 2012 at 10:48 AM, Vbvbrj <vbvbrj at gmail.com> wrote:
>>> Hello. I use uuid_broadcast to play a file to some leg:
>>> api:executeString("uuid_broadcast "..caller_uuid.."
>>> phrase::queue-position,"..pos.." aleg")
>>> But this pauses the moh file which is played to member waiting in
>>> callcenter. After two broadcasts, there is an error about leaking stream
>>> and adjusting of moh play. Is there a command to do the same but mux
>>> this playback of position with moh playing?
>
> Yes. But a caller hears music jump, which is not very interesting to let
> the users hear that, as he/she may think that the pbx is broken.
>
> Anyway, this is a suggestion that some one will be willing to implement.
> For now I use it as it is.
>
> Thank you.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

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

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

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


From mgg at giagnocavo.net  Fri Aug 31 05:19:31 2012
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Fri, 31 Aug 2012 01:19:31 +0000
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <0D1C698866F66045A6201FD0F59CAC90014678ACA7@EX.frontier.local>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
	<CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
	<CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
	<CAA8_NKAnX97E0dV7tpakg+wO-TzWdKm5wrm0qNP1EqODAVRoNw@mail.gmail.com>
	<CAPNTnnpHzcUzJ8Jnkj6Nma_2pZVyVhfWDA4RjBtJkuNWEMOdvg@mail.gmail.com>
	<0D1C698866F66045A6201FD0F59CAC90014678ACA7@EX.frontier.local>
Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FC3248@BLUPRD0711MB413.namprd07.prod.outlook.com>

At least on one of our boxes, running on CentOS 6.2 ends up with one FS thread stuck at full CPU. We haven't noticed anything getting worse as more load gets added. My feeling is that perhaps there's some sleep or something that is now much more granular with a tickless kernel so now it's waking up all the time, instead of 1000Hz. But several folks have told me tickles ain't the issue.

At any rate, machines are big and profitable, so losing a single core to a runaway thread isn't a huge priority for me :\.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Colin Mason
Sent: Thursday, August 30, 2012 4:17 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] RedHat 6.X Performance

What exactly is 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:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Guillermo Ruiz Camauer
Sent: Thursday, August 30, 2012 6:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] RedHat 6.X Performance

Is anyone having the same problems with RedHat or is this CentOS specific?

On Thu, Aug 30, 2012 at 6:23 PM, Gavin Henry <gavin.henry at gmail.com<mailto:gavin.henry at gmail.com>> wrote:
> Centos5 worked well for me, Centos6 worked for crap... I went to debian 6
> (squeeze) and problems went a way... using the latest stable on the debian6
> line installing from the netinstall (non-free variant as I have hardware
> thats not supported under the main netinstall iso) and then installing
> freeswitch via the instructions on the wiki and no problems so far
Same here. We're CentOS 5 here and Debian Squeezy for new boxes.

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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
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From mgg at giagnocavo.net  Fri Aug 31 05:19:30 2012
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Fri, 31 Aug 2012 01:19:30 +0000
Subject: [Freeswitch-users] Best way to stream record session?
In-Reply-To: <000001cd86a8$75efcd10$61cf6730$@207me.com>
References: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FC202C@BLUPRD0711MB413.namprd07.prod.outlook.com>
	<000001cd86a8$75efcd10$61cf6730$@207me.com>
Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FC3241@BLUPRD0711MB413.namprd07.prod.outlook.com>

Oh wow. I had seen mod_shout but thought it was purely playing back streams not also creating them. Thanks!

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stephen Dame
Sent: Thursday, August 30, 2012 6:11 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Best way to stream record session?

Mod_shout  configured  to an icecast server works well for this.   The wiki
has examples.

The latest freeswitch cookbook, also has section on telecast? Where you access call from http served directly from freeswitch.  Not sure how this scales if you have bunch of listeners.

Regards,
Stephen

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Giagnocavo
Sent: Thursday, August 30, 2012 3:04 AM
To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)
Subject: [Freeswitch-users] Best way to stream record session?

Hi all,

	I want to record and stream a (bridged) call to a non-channel endpoint. Sorta like mod_spy, but targeting a TCP or UDP connection or even a one-way RTP stream. What's the best way to approach this? What about using normal session record to a file, then "streaming" that file from an external program? Is FS guaranteed to write to the wav file regularly (like every second or so) in a manner that another program can keep reading? Or should I specify a file like "/dev/udp/1.2.3.4/666"?

	End goal is to have a separate machine that can provide a service like Internet radio and have it tap into live calls.

Cheers,
Michael



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 gvvsubhashkumar at gmail.com  Fri Aug 31 06:28:10 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Fri, 31 Aug 2012 07:58:10 +0530
Subject: [Freeswitch-users] DialPlan Generation
In-Reply-To: <CAKzWOxU9Tkpe2JMB_gUb5P2UneScyTkHaxzpy33R_RFHdH=sRQ@mail.gmail.com>
References: <CAFvodxXx0nN8kfg5V-E7M3oTVC34Sr_RJ-EgxzULQELLyZ+6tA@mail.gmail.com>
	<CAKzWOxU9Tkpe2JMB_gUb5P2UneScyTkHaxzpy33R_RFHdH=sRQ@mail.gmail.com>
Message-ID: <CAFvodxUJxDM0Soy3a2uLUZZ-M=iCRh+E_Xv9dA0baGBAoX=VqA@mail.gmail.com>

Thanks for the reply.

I have a question ,can freeswitch act as a proxy?


Thanks,
Subhash.


On Fri, Aug 31, 2012 at 1:07 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, Aug 30, 2012 at 9:59 AM, Subhash <gvvsubhashkumar at gmail.com>wrote:
>
>> Hi,
>>
>>      Which is the best way to create(static or dynamic) dialplan?
>>
>>     Please suggest me.
>>
>> The larger and more active your system becomes, the more valuable a
> dynamic dialplan (and other configs) become. As to how you do the dynamic
> stuff, that's really up to you. Using mod_xml_curl is a great way to make
> it very scalable and have fallback options.
> -MC
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 krice at freeswitch.org  Fri Aug 31 07:45:28 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 30 Aug 2012 22:45:28 -0500
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B612FC3248@BLUPRD0711MB413.namprd07.prod.outlook.com>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
	<CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
	<CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
	<CAA8_NKAnX97E0dV7tpakg+wO-TzWdKm5wrm0qNP1EqODAVRoNw@mail.gmail.com>
	<CAPNTnnpHzcUzJ8Jnkj6Nma_2pZVyVhfWDA4RjBtJkuNWEMOdvg@mail.gmail.com>
	<0D1C698866F66045A6201FD0F59CAC90014678ACA7@EX.frontier.local>
	<63B00DD1DA6A364E9F64A3A0BD2FE7B612FC3248@BLUPRD0711MB413.namprd07.prod.outlook.com>
Message-ID: <CAPcNkddX0xXdJsYm+UTwy=xo5Saw5gxk=5XF_rMcJi7yTZ+mzQ@mail.gmail.com>

I'm running probably one of the most stripped down FreeSWITCH configs you
can run ... Sofia only, bypass media, with a custom C routing module that
uses libpq directly...

the problem happens at somewhere around 50 to 100 CPS, system % goes thru
the roof, and its not IO.. loglevel 0... etc... now its also worth
mentioning that on 6.2 the number of context switches for similar amounts
of calls on the same physical hardware seems to be double vs something like
cent5 or deb6....

we're currently trying to get people to submit reports from oprofile so we
can isolate the issue... if it were just me, I would chalk it up to
something specific on the installation, but , the number of reporters with
similar observations is something to take notice of...


On Thu, Aug 30, 2012 at 8:19 PM, Michael Giagnocavo <mgg at giagnocavo.net>wrote:

>  At least on one of our boxes, running on CentOS 6.2 ends up with one FS
> thread stuck at full CPU. We haven?t noticed anything getting worse as more
> load gets added. My feeling is that perhaps there?s some sleep or something
> that is now much more granular with a tickless kernel so now it?s waking up
> all the time, instead of 1000Hz. But several folks have told me tickles
> ain?t the issue. ****
>
> ** **
>
> At any rate, machines are big and profitable, so losing a single core to a
> runaway thread isn?t a huge priority for me :\.****
>
> ** **
>
> -Michael****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Colin Mason
> *Sent:* Thursday, August 30, 2012 4:17 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] RedHat 6.X Performance****
>
>  ** **
>
> What exactly is the problem?****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Guillermo
> Ruiz Camauer
> *Sent:* Thursday, August 30, 2012 6:13 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] RedHat 6.X Performance****
>
> ** **
>
> Is anyone having the same problems with RedHat or is this CentOS specific?
> ****
>
> ** **
>
> On Thu, Aug 30, 2012 at 6:23 PM, Gavin Henry <gavin.henry at gmail.com>
> wrote:****
>
> > Centos5 worked well for me, Centos6 worked for crap... I went to debian 6
> > (squeeze) and problems went a way... using the latest stable on the
> debian6
> > line installing from the netinstall (non-free variant as I have hardware
> > thats not supported under the main netinstall iso) and then installing
> > freeswitch via the instructions on the wiki and no problems so far****
>
> Same here. We're CentOS 5 here and Debian Squeezy for new boxes.
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.surevoip.co.uk****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
>
>
> ****
>
> ** **
>
> --
> Guillermo Ruiz Camauer****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Fri Aug 31 07:49:20 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 30 Aug 2012 20:49:20 -0700 (PDT)
Subject: [Freeswitch-users] Problem with originated calls hanging in
	CS_SOFT_EXEC state
In-Reply-To: <5C0B6578-49A5-4174-87B4-C06E409EB341@vocalspace.com>
References: <FB1F0348-B48D-44FA-821D-1473E4EEDCE1@vocalspace.com>
	<5C0B6578-49A5-4174-87B4-C06E409EB341@vocalspace.com>
Message-ID: <1346384960442-7582404.post@n2.nabble.com>

call session.destroy() and report back



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-originated-calls-hanging-in-CS-SOFT-EXEC-state-tp7582377p7582404.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From cmason at frontiernetworks.ca  Fri Aug 31 07:54:13 2012
From: cmason at frontiernetworks.ca (Colin Mason)
Date: Thu, 30 Aug 2012 23:54:13 -0400
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAPcNkddX0xXdJsYm+UTwy=xo5Saw5gxk=5XF_rMcJi7yTZ+mzQ@mail.gmail.com>
Message-ID: <0D1C698866F66045A6201FD0F59CAC900145D57CF2@EX.frontier.local>

I do 250 cps and somewhere around 100000 sessions per day on a cent6 + fs1.2 install. The box has 8 cores and I never see it using more than 1 core. Is this worth oprofiling? If so, where/who should I send the oprofile to?

Colin

From: Ken Rice [mailto:krice at freeswitch.org]
Sent: Thursday, August 30, 2012 11:45 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] RedHat 6.X Performance

I'm running probably one of the most stripped down FreeSWITCH configs you can run ... Sofia only, bypass media, with a custom C routing module that uses libpq directly...

the problem happens at somewhere around 50 to 100 CPS, system % goes thru the roof, and its not IO.. loglevel 0... etc... now its also worth mentioning that on 6.2 the number of context switches for similar amounts of calls on the same physical hardware seems to be double vs something like cent5 or deb6....

we're currently trying to get people to submit reports from oprofile so we can isolate the issue... if it were just me, I would chalk it up to something specific on the installation, but , the number of reporters with similar observations is something to take notice of...


On Thu, Aug 30, 2012 at 8:19 PM, Michael Giagnocavo <mgg at giagnocavo.net<mailto:mgg at giagnocavo.net>> wrote:
At least on one of our boxes, running on CentOS 6.2 ends up with one FS thread stuck at full CPU. We haven?t noticed anything getting worse as more load gets added. My feeling is that perhaps there?s some sleep or something that is now much more granular with a tickless kernel so now it?s waking up all the time, instead of 1000Hz. But several folks have told me tickles ain?t the issue.

At any rate, machines are big and profitable, so losing a single core to a runaway thread isn?t a huge priority for me :\.

-Michael

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 Colin Mason
Sent: Thursday, August 30, 2012 4:17 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] RedHat 6.X Performance

What exactly is 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:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Guillermo Ruiz Camauer
Sent: Thursday, August 30, 2012 6:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] RedHat 6.X Performance

Is anyone having the same problems with RedHat or is this CentOS specific?

On Thu, Aug 30, 2012 at 6:23 PM, Gavin Henry <gavin.henry at gmail.com<mailto:gavin.henry at gmail.com>> wrote:
> Centos5 worked well for me, Centos6 worked for crap... I went to debian 6
> (squeeze) and problems went a way... using the latest stable on the debian6
> line installing from the netinstall (non-free variant as I have hardware
> thats not supported under the main netinstall iso) and then installing
> freeswitch via the instructions on the wiki and no problems so far
Same here. We're CentOS 5 here and Debian Squeezy for new boxes.

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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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



--
Guillermo Ruiz Camauer

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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


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From govoiper at gmail.com  Fri Aug 31 09:24:01 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 31 Aug 2012 10:24:01 +0500
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <8D066FE8-DB24-4928-8508-317E23B858EA@me.com>
References: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
	<8D066FE8-DB24-4928-8508-317E23B858EA@me.com>
Message-ID: <CAJUJwthxmhd0u6G35ANgiCzdHv1KY-qKK5ZD_Xx_+S-cGYMXVg@mail.gmail.com>

Hi,
Can you please manage a proper sip trace for this ! I'd really love to
troubleshoot and analyse the sip traces.
What is the equipment on your A-leg ?

BR
Sammy



On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame at me.com>wrote:

> I would like to add an update to this and maybe someone has some other
> suggestions - It does look like that FS is disconnecting the call with
> cause code 31  [NORMAL_UNSPECIFIED] I did not notice the time stamp
> difference on the Bye when I first looked at the log's
>
> The basis of the issue still stands a 491 Results in the B-Leg being
> disconnected
>
> On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame at me.com>
> wrote:
>
> We are seeing some instances when we send a invite from the B-Leg back to
> FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has
> not fully completed yet causing a return of a 491 from the A-Leg side
> causing the call to be disconnected. wanted to see if anyone else has seen
> something like this while running FS and if anyone had any suggestions on a
> fix?
>
> A-Leg
> Invite into Freeswitch
> 100 Trying back from FS to A-Leg
> 180 Ringing from FS to A-Leg
> 200 OK from FS to A-Leg at 15:08:14.638799
> Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515
> 100 Giving a try form A-Leg to FS at 15:08:14.749757
> 491 From A-Leg to FS at 15:08:14.780968
> ACK from FS to A-Leg at 15:08:14.781102
> ACK from FS to A-Leg at 15:08:14.797143
> BYE from A-LEG to FS 15:11:10.791963
>
>    481 Call Does Not Exist back to A-LEG
>
>
> B-Leg
> Invite from Freeswitch to B-Leg
> 100 Giving a try from B-Leg
> 180 Ringing from B-Leg
> 200 OK  from B-Leg at 15:08:14.635670
> ACK from FS to B-Leg  at 15:08:14.637044
> Invite from B-Leg to FS at 15:08:14.748623
> 100 Trying from FS to B-Leg at 15:08:14.748954
> 491 from FS to B-Leg at 15:08:14.782169
> ACK from B-Leg to FS at 15:08:14.782372
> BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mike.burlingame at me.com  Fri Aug 31 09:29:59 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Thu, 30 Aug 2012 22:29:59 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <CAJUJwthxmhd0u6G35ANgiCzdHv1KY-qKK5ZD_Xx_+S-cGYMXVg@mail.gmail.com>
References: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
	<8D066FE8-DB24-4928-8508-317E23B858EA@me.com>
	<CAJUJwthxmhd0u6G35ANgiCzdHv1KY-qKK5ZD_Xx_+S-cGYMXVg@mail.gmail.com>
Message-ID: <9065DE2C-DF4A-4E6C-B436-1A83B2AFA664@me.com>

Sure I will put logs and sip dump up tomorrow A-Leg == Sonus B-Leg == Broadsoft M6

Sent from my iPhone 4S

On Aug 30, 2012, at 10:24 PM, SamyGo <govoiper at gmail.com> wrote:

> Hi,
> Can you please manage a proper sip trace for this ! I'd really love to troubleshoot and analyse the sip traces.
> What is the equipment on your A-leg ? 
> 
> BR
> Sammy
> 
>  
> 
> On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> I would like to add an update to this and maybe someone has some other suggestions - It does look like that FS is disconnecting the call with cause code 31  [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on the Bye when I first looked at the log's
>> 
>> The basis of the issue still stands a 491 Results in the B-Leg being disconnected
>> 
>> On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> 
>>> We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
>>> 
>>> A-Leg
>>> Invite into Freeswitch
>>> 100 Trying back from FS to A-Leg
>>> 180 Ringing from FS to A-Leg
>>> 200 OK from FS to A-Leg at 15:08:14.638799
>>> Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515
>>> 100 Giving a try form A-Leg to FS at 15:08:14.749757
>>> 491 From A-Leg to FS at 15:08:14.780968
>>> ACK from FS to A-Leg at 15:08:14.781102
>>> ACK from FS to A-Leg at 15:08:14.797143
>>> BYE from A-LEG to FS 15:11:10.791963
>>    481 Call Does Not Exist back to A-LEG
>>> 
>>> B-Leg
>>> Invite from Freeswitch to B-Leg
>>> 100 Giving a try from B-Leg
>>> 180 Ringing from B-Leg
>>> 200 OK  from B-Leg at 15:08:14.635670
>>> ACK from FS to B-Leg  at 15:08:14.637044
>>> Invite from B-Leg to FS at 15:08:14.748623
>>> 100 Trying from FS to B-Leg at 15:08:14.748954
>>> 491 from FS to B-Leg at 15:08:14.782169
>>> ACK from B-Leg to FS at 15:08:14.782372
>>> BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 khuenm at vega.com.vn  Fri Aug 31 08:02:37 2012
From: khuenm at vega.com.vn (Khue Nguyen Minh)
Date: Fri, 31 Aug 2012 11:02:37 +0700
Subject: [Freeswitch-users] No Dialplan on answer channel
Message-ID: <CAO-yT6hxe8eCcuT36WEjqmLDKYYfyGHnv3c2O_Opz3q6UMOhmA@mail.gmail.com>

Hi all,

I want make outbound call to a user with command:

*originate sofia/external/1000 at somewhere.com '&javascript(hello.js
/opt/freeswitch/sounds/2.mp3)'*

my javascript will play file /opt/freeswitch/sounds/2.mp3.

The call make successful but hangup immediately. I see log in fs_cli and
see this line "No Dialplan on answer channel". How I can fix this problem?

Thanks
Khue Nguyen.
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From msc at freeswitch.org  Fri Aug 31 09:43:17 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 22:43:17 -0700
Subject: [Freeswitch-users] DialPlan Generation
In-Reply-To: <CAFvodxUJxDM0Soy3a2uLUZZ-M=iCRh+E_Xv9dA0baGBAoX=VqA@mail.gmail.com>
References: <CAFvodxXx0nN8kfg5V-E7M3oTVC34Sr_RJ-EgxzULQELLyZ+6tA@mail.gmail.com>
	<CAKzWOxU9Tkpe2JMB_gUb5P2UneScyTkHaxzpy33R_RFHdH=sRQ@mail.gmail.com>
	<CAFvodxUJxDM0Soy3a2uLUZZ-M=iCRh+E_Xv9dA0baGBAoX=VqA@mail.gmail.com>
Message-ID: <CAKzWOxUfsgTc4dLTRktkY4kQyZCnBuUyYLcK_3eznwJDJt3uNA@mail.gmail.com>

On Thu, Aug 30, 2012 at 7:28 PM, Subhash <gvvsubhashkumar at gmail.com> wrote:

> Thanks for the reply.
>
> I have a question ,can freeswitch act as a proxy?
>
It can do a few proxy-ish things, but really it's a B2BUA.
-MC
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From gvvsubhashkumar at gmail.com  Fri Aug 31 09:50:13 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Thu, 30 Aug 2012 22:50:13 -0700
Subject: [Freeswitch-users] DialPlan Generation
In-Reply-To: <CAKzWOxUfsgTc4dLTRktkY4kQyZCnBuUyYLcK_3eznwJDJt3uNA@mail.gmail.com>
References: <CAFvodxXx0nN8kfg5V-E7M3oTVC34Sr_RJ-EgxzULQELLyZ+6tA@mail.gmail.com>
	<CAKzWOxU9Tkpe2JMB_gUb5P2UneScyTkHaxzpy33R_RFHdH=sRQ@mail.gmail.com>
	<CAFvodxUJxDM0Soy3a2uLUZZ-M=iCRh+E_Xv9dA0baGBAoX=VqA@mail.gmail.com>
	<CAKzWOxUfsgTc4dLTRktkY4kQyZCnBuUyYLcK_3eznwJDJt3uNA@mail.gmail.com>
Message-ID: <CAFvodxXh48bvkG8=N9EXfxFGZcFvz54050XcMvJON0ku+56TkQ@mail.gmail.com>

Where can i find the list of the things that freeswitch can do as a proxy?


Thanks,
Subhash.


On Thu, Aug 30, 2012 at 10:43 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Thu, Aug 30, 2012 at 7:28 PM, Subhash <gvvsubhashkumar at gmail.com>wrote:
>
>> Thanks for the reply.
>>
>> I have a question ,can freeswitch act as a proxy?
>>
> It can do a few proxy-ish things, but really it's a B2BUA.
> -MC
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 Aug 31 09:53:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 22:53:27 -0700
Subject: [Freeswitch-users] No Dialplan on answer channel
In-Reply-To: <CAO-yT6hxe8eCcuT36WEjqmLDKYYfyGHnv3c2O_Opz3q6UMOhmA@mail.gmail.com>
References: <CAO-yT6hxe8eCcuT36WEjqmLDKYYfyGHnv3c2O_Opz3q6UMOhmA@mail.gmail.com>
Message-ID: <CAKzWOxXyYZH8UUSDEVzvcF+avzFLYQaT-6sNbvsozSiW-FmsJQ@mail.gmail.com>

It might be the location of your single quotes. Try something a little
different, like this:

*originate sofia/external/1000 at somewhere.com &javascript('hello.js
/opt/freeswitch/sounds/2.mp3')*

If that doesn't work then keep tinkering. If you still need more help then
I recommend putting your console output into pastebin.freeswitch.org and
linking back here in this thread. Be sure to use "FreeSWITCH Log" as the
syntax highlighting.

-MC

On Thu, Aug 30, 2012 at 9:02 PM, Khue Nguyen Minh <khuenm at vega.com.vn>wrote:

> Hi all,
>
> I want make outbound call to a user with command:
>
> *originate sofia/external/1000 at somewhere.com '&javascript(hello.js
> /opt/freeswitch/sounds/2.mp3)'*
>
> my javascript will play file /opt/freeswitch/sounds/2.mp3.
>
> The call make successful but hangup immediately. I see log in fs_cli and
> see this line "No Dialplan on answer channel". How I can fix this problem?
>
> Thanks
> Khue Nguyen.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Fri Aug 31 10:00:32 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 23:00:32 -0700
Subject: [Freeswitch-users] uuid_broadcast muxing.
In-Reply-To: <CAKbxfG92Qiw54Grik6mqEP4+GM6+WNN=a2TEuDx-a94+RDdd0A@mail.gmail.com>
References: <503E39C6.40306@gmail.com>
	<CAKbxfG9CsU4U+xEYmVBWrGdXDDdsSfuXQEwQW7HP5bJQtogLGA@mail.gmail.com>
	<503EF955.1080400@gmail.com>
	<CAKbxfG92Qiw54Grik6mqEP4+GM6+WNN=a2TEuDx-a94+RDdd0A@mail.gmail.com>
Message-ID: <CAKzWOxVJRbE4q0a0R7tDWMc1qYfYObxE055DTT-YmLEgyJGXbw@mail.gmail.com>

FYI, for posterity's sake I added this thread and Tony's response to the
local_stream wiki page's FAQ section:
http://wiki.freeswitch.org/wiki/Mod_local_stream#What.27s_this_I_keep_seeing_about_a_leaking_stream_handle.3F

-MC

On Thu, Aug 30, 2012 at 6:19 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> nobody is actually hearing it discard the audio its only the audio to
> the handle where you have the music paused.
> A normal file handle waits for you to read, the local_stream one has a
> buffer feeding it from a master stream, so when you don't read from
> it, you lose the audio because the music is playing on without you
> even when you pause.
>
>
> On Thu, Aug 30, 2012 at 12:25 AM, Vbvbrj <vbvbrj at gmail.com> wrote:
> > On 30.08.2012 02:21, Anthony Minessale wrote:
> >> The error is harmless warning to remind you the local_stream is being
> ignored.
> >>
> >>
> >> On Wed, Aug 29, 2012 at 10:48 AM, Vbvbrj <vbvbrj at gmail.com> wrote:
> >>> Hello. I use uuid_broadcast to play a file to some leg:
> >>> api:executeString("uuid_broadcast "..caller_uuid.."
> >>> phrase::queue-position,"..pos.." aleg")
> >>> But this pauses the moh file which is played to member waiting in
> >>> callcenter. After two broadcasts, there is an error about leaking
> stream
> >>> and adjusting of moh play. Is there a command to do the same but mux
> >>> this playback of position with moh playing?
> >
> > Yes. But a caller hears music jump, which is not very interesting to let
> > the users hear that, as he/she may think that the pbx is broken.
> >
> > Anyway, this is a suggestion that some one will be willing to implement.
> > For now I use it as it is.
> >
> > Thank you.
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From vbvbrj at gmail.com  Fri Aug 31 10:05:55 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Fri, 31 Aug 2012 09:05:55 +0300
Subject: [Freeswitch-users] uuid_broadcast muxing.
In-Reply-To: <CAKbxfG92Qiw54Grik6mqEP4+GM6+WNN=a2TEuDx-a94+RDdd0A@mail.gmail.com>
References: <503E39C6.40306@gmail.com>
	<CAKbxfG9CsU4U+xEYmVBWrGdXDDdsSfuXQEwQW7HP5bJQtogLGA@mail.gmail.com>
	<503EF955.1080400@gmail.com>
	<CAKbxfG92Qiw54Grik6mqEP4+GM6+WNN=a2TEuDx-a94+RDdd0A@mail.gmail.com>
Message-ID: <50405443.4040201@gmail.com>

On 31.08.2012 04:19, Anthony Minessale wrote:
> nobody is actually hearing it discard the audio its only the audio to
> the handle where you have the music paused.
> A normal file handle waits for you to read, the local_stream one has a
> buffer feeding it from a master stream, so when you don't read from
> it, you lose the audio because the music is playing on without you
> even when you pause.

Anthony, if we take the audio from moh from git, it is hard to hear it 
was missed. But if to take some other music with louder sound, or from 
shoutcast, it will be heard audio missing and jumping. It will be great 
to have mux option for uuid_broadcast.

Mimiko desu.



From gabe at gundy.org  Fri Aug 31 10:08:51 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 31 Aug 2012 00:08:51 -0600
Subject: [Freeswitch-users] Asking a Question (a.k.a. Gabe must be getting
	old)
Message-ID: <CAMwZ-tJ7Asfd_MSWqNjCdEsSF=DOx9daGZ_T5CbWzoxfTb2UfA@mail.gmail.com>

This should be required reading before posting questions to the list:

http://www.catb.org/esr/faqs/smart-questions.html#intro

Special thanks to those people on the list who take special care to
ask *smart* questions and those who provide valuable answers. You know
who you are ;)


Gabe


From msc at freeswitch.org  Fri Aug 31 10:11:12 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 30 Aug 2012 23:11:12 -0700
Subject: [Freeswitch-users] DialPlan Generation
In-Reply-To: <CAFvodxXh48bvkG8=N9EXfxFGZcFvz54050XcMvJON0ku+56TkQ@mail.gmail.com>
References: <CAFvodxXx0nN8kfg5V-E7M3oTVC34Sr_RJ-EgxzULQELLyZ+6tA@mail.gmail.com>
	<CAKzWOxU9Tkpe2JMB_gUb5P2UneScyTkHaxzpy33R_RFHdH=sRQ@mail.gmail.com>
	<CAFvodxUJxDM0Soy3a2uLUZZ-M=iCRh+E_Xv9dA0baGBAoX=VqA@mail.gmail.com>
	<CAKzWOxUfsgTc4dLTRktkY4kQyZCnBuUyYLcK_3eznwJDJt3uNA@mail.gmail.com>
	<CAFvodxXh48bvkG8=N9EXfxFGZcFvz54050XcMvJON0ku+56TkQ@mail.gmail.com>
Message-ID: <CAKzWOxXg18OxndmMeSUGPVk_6nuC1H95Rw864KyJqWA7V_BgUQ@mail.gmail.com>

On Thu, Aug 30, 2012 at 10:50 PM, Subhash <gvvsubhashkumar at gmail.com> wrote:

> Where can i find the list of the things that freeswitch can do as a proxy?
>
> There is no list per se. About the only way to find out is to google
lists.freeswitch.org with your specific inquiry and see if it has been
discussed before.

Your questions are both broad and vague. Is there any chance you could add
some detail? What are you using FreeSWITCH for? It may be that FreeSWITCH
is not the right tool, or at least not the only tool you need. A proxy and
a B2BUA go hand-in-hand, each serving a specific purpose. There are lots of
knowledgeable people who can answer specific questions, but that means a
little more work on your part. I hope that sounds like a reasonable
tradeoff. :)

-MC


>
> Thanks,
> Subhash.
>
>
> On Thu, Aug 30, 2012 at 10:43 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Thu, Aug 30, 2012 at 7:28 PM, Subhash <gvvsubhashkumar at gmail.com>wrote:
>>
>>> Thanks for the reply.
>>>
>>> I have a question ,can freeswitch act as a proxy?
>>>
>> It can do a few proxy-ish things, but really it's a B2BUA.
>> -MC
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From miha at softnet.si  Fri Aug 31 10:32:26 2012
From: miha at softnet.si (Miha)
Date: Fri, 31 Aug 2012 08:32:26 +0200
Subject: [Freeswitch-users] group call pickup
In-Reply-To: <CAC-LwPOH_5nbbZ3Z6iM3JdkFrmZjP_NMBCjV7AB1tHux2GzhkA@mail.gmail.com>
References: <503E18D5.2020907@softnet.si>
	<CAC-LwPN4brMLWT5rxpUG--hiwUZz687SPq7C90yVf1BRZPQUJQ@mail.gmail.com>
	<503E2AF1.3030607@softnet.si>
	<CAC-LwPNHdfPwjpP5pkOOmvC7qWLuY4L288ca+LsqK9hZ5VvCyg@mail.gmail.com>
	<503E33C2.2090007@softnet.si> <503F07BF.7070205@softnet.si>
	<CAC-LwPOH_5nbbZ3Z6iM3JdkFrmZjP_NMBCjV7AB1tHux2GzhkA@mail.gmail.com>
Message-ID: <50405A7A.1030101@softnet.si>

HI Vik,

I did as you said, just replace $1 with 1 and still the same... I am 
using FS 1.2 rc2. Does any one know what I am missing?

2012-08-31 10:24:57.049318 [DEBUG] sofia.c:6274 
(sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State Change 
CS_NEW -> CS_INIT
2012-08-31 10:24:57.049318 [DEBUG] switch_core_session.c:1229 Send 
signal sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx [BREAK]
2012-08-31 10:24:57.049318 [DEBUG] switch_core_state_machine.c:385 
(sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) Running State 
Change CS_INIT
2012-08-31 10:24:57.049318 [DEBUG] switch_core_state_machine.c:424 
(sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State INIT
2012-08-31 10:24:57.049318 [DEBUG] mod_sofia.c:85 
sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx SOFIA INIT
2012-08-31 10:24:57.049318 [DEBUG] mod_sofia.c:125 
(sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State Change 
CS_INIT -> CS_ROUTING
2012-08-31 10:24:57.049318 [DEBUG] switch_core_session.c:1229 Send 
signal sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx [BREAK]
2012-08-31 10:24:57.049318 [DEBUG] switch_core_state_machine.c:424 
(sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State INIT going 
to sleep
2012-08-31 10:24:57.049318 [DEBUG] switch_core_state_machine.c:385 
(sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) Running State 
Change CS_ROUTING
2012-08-31 10:24:57.049318 [DEBUG] switch_channel.c:1934 
(sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) Callstate Change 
DOWN -> RINGING
2012-08-31 10:24:57.049318 [DEBUG] switch_core_state_machine.c:433 
(sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State ROUTING
2012-08-31 10:24:57.049318 [DEBUG] mod_sofia.c:148 
sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx SOFIA ROUTING
2012-08-31 10:24:57.049318 [DEBUG] switch_core_state_machine.c:104 
sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Standard ROUTING
2012-08-31 10:24:57.049318 [INFO] mod_dialplan_xml.c:485 Processing 
018108753 <018108753.enterprise>->*572 in context xxx.xxx.xxx.xxx
Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx parsing 
[xxx.xxx.xxx.xxx->group-intercept-number] continue=false
Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Regex 
(FAIL) [group-intercept-number] destination_number(*572) =~ /^\*6(\d+)$/ 
break=on-false
Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx parsing 
[xxx.xxx.xxx.xxx->group-intercept] continue=false
Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Regex 
(PASS) [group-intercept] destination_number(*572) =~ /^\*57(\d+)$/ 
break=on-false
Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Action 
pickup(1)
2012-08-31 10:24:57.049318 [DEBUG] switch_core_state_machine.c:154 
(sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State Change 
CS_ROUTING -> CS_EXECUTE
2012-08-31 10:24:57.049318 [DEBUG] switch_core_session.c:1229 Send 
signal sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx [BREAK]
2012-08-31 10:24:57.049318 [DEBUG] switch_core_state_machine.c:433 
(sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State ROUTING 
going to sleep
2012-08-31 10:24:57.049318 [DEBUG] switch_core_state_machine.c:385 
(sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) Running State 
Change CS_EXECUTE
2012-08-31 10:24:57.049318 [DEBUG] switch_core_state_machine.c:440 
(sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State EXECUTE
2012-08-31 10:24:57.049318 [DEBUG] mod_sofia.c:241 
sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx SOFIA EXECUTE
2012-08-31 10:24:57.049318 [DEBUG] switch_core_state_machine.c:196 
sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Standard EXECUTE
EXECUTE sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx pickup(1)
2012-08-31 10:24:57.049318 [NOTICE] switch_core_state_machine.c:249 
sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx has executed the 
last dialplan instruction, hanging up.
2012-08-31 10:24:57.049318 [DEBUG] switch_channel.c:2914 
(sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) Callstate Change 
RINGING -> HANGUP
2012-08-31 10:24:57.049318 [NOTICE] switch_core_state_machine.c:251 
Hangup sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx [CS_EXECUTE] 
[NORMAL_CLEARING]
2012-08-31 10:24:57.049318 [DEBUG] switch_channel.c:2937 Send signal 
sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx [KILL]
2012-08-31 10:24:57.049318 [DEBUG] switch_core_session.c:1229 Send 
signal sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx [BREAK]

Thanks!
Miha

On 8/30/2012 2:13 PM, Vik Killa wrote:
> I'm not sure where the problem is here... try doing something simple like this:
> <action application="bridge"
> data="leg_timeout=30]user/${sip_to_user}.enterprise at xxx.xxx.xxx.xxx|user/${call_forwarding_number__nore}.enterprise at xxx.xxx.xxx.xxx,pickup/1"/>
>
> and this:
> <extension name="group_pickup">
>     <condition field="destination_number" expression="^\*57(\d+)$">
>       <action application="pickup" data="1"/>
>     </condition>
> </extension>
>
>
>
> On Thu, Aug 30, 2012 at 2:27 AM, Miha <miha at softnet.si> wrote:
>> Hi,
>>
>> I also tried like it is written on wiki, with numberic groups. But still
>> group pickup/call intercept do not work for me.
>>
>> action application="bridge"
>> data="[leg_timeout=30]user/${sip_to_user}.enterprise at xxx.xxx.xxx.xxx|user/${call_forwarding_number__nore}.enterprise at xxx.xxx.xxx.xxx,
>> pickup/1"/>
>>
>> <extension name="group_pickup">
>>     <condition field="destination_number" expression="^\*57(\d+)$">
>>       <action application="pickup" data="$1"/>
>>     </condition>
>> </extension>
>>
>> 2012-08-30 10:16:56.189010 [INFO] mod_dialplan_xml.c:485 Processing 018108753 <018108753.enterprise>->*571 in context xxx.xxx.xxx.xxx
>> Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx parsing [xxx.xxx.xxx.xxx->group-intercept-number] continue=false
>> Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Regex (FAIL) [group-intercept-number] destination_number(*571) =~ /^\*6(\d+)$/ break=on-false
>> Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx parsing [xxx.xxx.xxx.xxx->group-intercept] continue=false
>> Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Regex (PASS) [group-intercept] destination_number(*571) =~ /^\*57(\d+)$/ break=on-false
>> Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Action pickup(1)
>> 2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State Change CS_ROUTING -> CS_EXECUTE
>> 2012-08-30 10:16:56.189010 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx [BREAK]
>> 2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State ROUTING going to sleep
>> 2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) Running State Change CS_EXECUTE
>> 2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx) State EXECUTE
>> 2012-08-30 10:16:56.189010 [DEBUG] mod_sofia.c:241 sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx SOFIA EXECUTE
>> 2012-08-30 10:16:56.189010 [DEBUG] switch_core_state_machine.c:196 sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Standard EXECUTE
>> EXECUTE sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx pickup(1)
>>
>>
>>
>>
>>
>> On 8/29/2012 5:22 PM, Miha wrote:
>>> @Vik,
>>>
>>> sorry for my stupid questions...
>>>
>>> It is matched. As I am trying with account that are in the same callgroup.
>>>
>>> log:
>>>
>>> EXECUTE
>>> sofia/internal/018108754.enterprise at xxx.xxx.xxx.xxxbridge([leg_timeout=30]user/018108755.enterprise at xxx.xxx.xxx.xxx|user/.enterprise at xxx.xxx.xxx.xxx,
>>> pickup/test)
>>>
>>> afer *5:
>>>
>>> Dialplan: sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx Action
>>> pickup(${callgroup})
>>>
>>> EXECUTE sofia/internal/018108753.enterprise at xxx.xxx.xxx.xxx pickup(test)
>>>
>>> thanks!
>>>
>>> Miha
>>>
>>> On 8/29/2012 5:02 PM, Vik Killa wrote:
>>>> pickup/XXXX only needs to match here
>>>> <action application="pickup" data="XXXX"/>
>>>>
>>>> You can use whatever variable as long as those two match.
>>>>
>>>> On Wed, Aug 29, 2012 at 10:45 AM, Miha <miha at softnet.si> wrote:
>>>>> Hi @Vik,
>>>>>
>>>>> I am having problem with understanding how this works. After I bridge
>>>>> data I put like this:
>>>>>
>>>>>      <anti-action application="bridge"
>>>>> data="[leg_timeout=30]user/${sip_to_user}.enterprise at xxx.xxx.xxx.xxx|user/${call_forwarding_number__nore}.enterprise at xxx.xxx.xxx.xxx,
>>>>> pickup/${callgroup
>>>>> }"/>
>>>>>
>>>>> To do pickup I am doing:
>>>>>
>>>>>
>>>>> <extension name="group-intercept">
>>>>>           <condition field="destination_number" expression="^\*5$">
>>>>> <action application="pickup" data="${groupcall}"/>
>>>>>              </condition>
>>>>>         </extension>
>>>>>
>>>>>
>>>>> What I am doing wrong or what I am missing?
>>>>>
>>>>> Thanks!
>>>>>
>>>>> Miha
>>>>> On 8/29/2012 3:49 PM, Vik Killa wrote:
>>>>>> That method of call pickup is out-dated, try using this, it works great:
>>>>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pickup
>>>>>>
>>>>>> On Wed, Aug 29, 2012 at 9:27 AM, Miha <miha at softnet.si> wrote:
>>>>>>> Hi,
>>>>>>>
>>>>>>> I have implemented call pickup, which is not working as it should.
>>>>>>>
>>>>>>> When call comes in, user from same group can pick it up, but problem appears
>>>>>>> in this scenario:
>>>>>>>
>>>>>>> A calls B. C do call pickup with *5 (in dialplan). OK this works (A is
>>>>>>> talking with C). But when D is calling E, if someone from same group pick
>>>>>>> this call, it do not pick D, it picks call A. Why? What I am doing wrong?
>>>>>>>
>>>>>>> I hope you understand what I mean:D
>>>>>>>
>>>>>>> my dialplan:
>>>>>>>
>>>>>>>      <action application="set"
>>>>>>> data="called_party_callgroup=${user_data(${destination_number}.enterprise at xxx.xxx.xxx.xxx
>>>>>>> var callgroup)}"/>
>>>>>>>                     <action application="hash"
>>>>>>> data="insert/${destination_number}/${called_party_callgroup}/${uuid}"/>
>>>>>>>                     <action application="hash"
>>>>>>> data="insert/last_dial/${called_party_callgroup}/${uuid}"/>
>>>>>>>
>>>>>>>
>>>>>>> intercept:
>>>>>>>
>>>>>>>
>>>>>>>            <extension name="group-intercept">
>>>>>>>           <condition field="destination_number" expression="^\*5$">
>>>>>>>             <action application="answer"/>
>>>>>>>              <action application="intercept"
>>>>>>> data="${hash(select/last_dial/${callgroup})}"/>
>>>>>>>
>>>>>>>             <action application="sleep" data="2000"/>
>>>>>>>           </condition>
>>>>>>>         </extension>
>>>>>>>
>>>>>>> Thanks!
>>>>>>> Miha
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 govoiper at gmail.com  Fri Aug 31 10:42:07 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 31 Aug 2012 11:42:07 +0500
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <9065DE2C-DF4A-4E6C-B436-1A83B2AFA664@me.com>
References: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
	<8D066FE8-DB24-4928-8508-317E23B858EA@me.com>
	<CAJUJwthxmhd0u6G35ANgiCzdHv1KY-qKK5ZD_Xx_+S-cGYMXVg@mail.gmail.com>
	<9065DE2C-DF4A-4E6C-B436-1A83B2AFA664@me.com>
Message-ID: <CAJUJwthU62H9o3iutVpt3EZO2-UKW0PXk6fJgFeGUO7NT1W0VQ@mail.gmail.com>

OK, and can you see whats happening inside the SONUS when it gets the
second invite (debug logs etc)? Can you control the M6 to stop re-invites !?

On Fri, Aug 31, 2012 at 10:29 AM, Mike Burlingame <mike.burlingame at me.com>wrote:

> Sure I will put logs and sip dump up tomorrow A-Leg == Sonus B-Leg ==
> Broadsoft M6
>
> Sent from my iPhone 4S
>
> On Aug 30, 2012, at 10:24 PM, SamyGo <govoiper at gmail.com> wrote:
>
> Hi,
> Can you please manage a proper sip trace for this ! I'd really love to
> troubleshoot and analyse the sip traces.
> What is the equipment on your A-leg ?
>
> BR
> Sammy
>
>
>
> On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame at me.com>wrote:
>
>> I would like to add an update to this and maybe someone has some other
>> suggestions - It does look like that FS is disconnecting the call with
>> cause code 31  [NORMAL_UNSPECIFIED] I did not notice the time stamp
>> difference on the Bye when I first looked at the log's
>>
>> The basis of the issue still stands a 491 Results in the B-Leg being
>> disconnected
>>
>> On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame at me.com>
>> wrote:
>>
>> We are seeing some instances when we send a invite from the B-Leg back to
>> FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has
>> not fully completed yet causing a return of a 491 from the A-Leg side
>> causing the call to be disconnected. wanted to see if anyone else has seen
>> something like this while running FS and if anyone had any suggestions on a
>> fix?
>>
>> A-Leg
>> Invite into Freeswitch
>> 100 Trying back from FS to A-Leg
>> 180 Ringing from FS to A-Leg
>> 200 OK from FS to A-Leg at 15:08:14.638799
>> Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515
>> 100 Giving a try form A-Leg to FS at 15:08:14.749757
>> 491 From A-Leg to FS at 15:08:14.780968
>> ACK from FS to A-Leg at 15:08:14.781102
>> ACK from FS to A-Leg at 15:08:14.797143
>> BYE from A-LEG to FS 15:11:10.791963
>>
>>    481 Call Does Not Exist back to A-LEG
>>
>>
>> B-Leg
>> Invite from Freeswitch to B-Leg
>> 100 Giving a try from B-Leg
>> 180 Ringing from B-Leg
>> 200 OK  from B-Leg at 15:08:14.635670
>> ACK from FS to B-Leg  at 15:08:14.637044
>> Invite from B-Leg to FS at 15:08:14.748623
>> 100 Trying from FS to B-Leg at 15:08:14.748954
>> 491 from FS to B-Leg at 15:08:14.782169
>> ACK from B-Leg to FS at 15:08:14.782372
>> BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mike.burlingame at me.com  Fri Aug 31 11:03:53 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 31 Aug 2012 00:03:53 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <CAJUJwthU62H9o3iutVpt3EZO2-UKW0PXk6fJgFeGUO7NT1W0VQ@mail.gmail.com>
References: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
	<8D066FE8-DB24-4928-8508-317E23B858EA@me.com>
	<CAJUJwthxmhd0u6G35ANgiCzdHv1KY-qKK5ZD_Xx_+S-cGYMXVg@mail.gmail.com>
	<9065DE2C-DF4A-4E6C-B436-1A83B2AFA664@me.com>
	<CAJUJwthU62H9o3iutVpt3EZO2-UKW0PXk6fJgFeGUO7NT1W0VQ@mail.gmail.com>
Message-ID: <AD70ED57-CDBB-4F9D-AE0F-77008F9EB45E@me.com>

The Sonus and M6 are out of my control they belong to my upstream and downstream so I have no access to them

Broadsoft support will not work this issue because FS ACKs the 200 OK so dialog is completed at least on the B-Leg so that's when the M6 sends the 2nd invite however the dialog has not completed on the A-Leg but FS still passes the Invite along resulting in the 491 

Sent from my iPhone 4S

On Aug 30, 2012, at 11:42 PM, SamyGo <govoiper at gmail.com> wrote:

> OK, and can you see whats happening inside the SONUS when it gets the second invite (debug logs etc)? Can you control the M6 to stop re-invites !?
> 
> On Fri, Aug 31, 2012 at 10:29 AM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> Sure I will put logs and sip dump up tomorrow A-Leg == Sonus B-Leg == Broadsoft M6
>> 
>> Sent from my iPhone 4S
>> 
>> On Aug 30, 2012, at 10:24 PM, SamyGo <govoiper at gmail.com> wrote:
>> 
>>> Hi,
>>> Can you please manage a proper sip trace for this ! I'd really love to troubleshoot and analyse the sip traces.
>>> What is the equipment on your A-leg ? 
>>> 
>>> BR
>>> Sammy
>>> 
>>>  
>>> 
>>> On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>> I would like to add an update to this and maybe someone has some other suggestions - It does look like that FS is disconnecting the call with cause code 31  [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on the Bye when I first looked at the log's
>>>> 
>>>> The basis of the issue still stands a 491 Results in the B-Leg being disconnected
>>>> 
>>>> On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>>> 
>>>>> We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
>>>>> 
>>>>> A-Leg
>>>>> Invite into Freeswitch
>>>>> 100 Trying back from FS to A-Leg
>>>>> 180 Ringing from FS to A-Leg
>>>>> 200 OK from FS to A-Leg at 15:08:14.638799
>>>>> Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515
>>>>> 100 Giving a try form A-Leg to FS at 15:08:14.749757
>>>>> 491 From A-Leg to FS at 15:08:14.780968
>>>>> ACK from FS to A-Leg at 15:08:14.781102
>>>>> ACK from FS to A-Leg at 15:08:14.797143
>>>>> BYE from A-LEG to FS 15:11:10.791963
>>>>    481 Call Does Not Exist back to A-LEG
>>>>> 
>>>>> B-Leg
>>>>> Invite from Freeswitch to B-Leg
>>>>> 100 Giving a try from B-Leg
>>>>> 180 Ringing from B-Leg
>>>>> 200 OK  from B-Leg at 15:08:14.635670
>>>>> ACK from FS to B-Leg  at 15:08:14.637044
>>>>> Invite from B-Leg to FS at 15:08:14.748623
>>>>> 100 Trying from FS to B-Leg at 15:08:14.748954
>>>>> 491 from FS to B-Leg at 15:08:14.782169
>>>>> ACK from B-Leg to FS at 15:08:14.782372
>>>>> BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From govoiper at gmail.com  Fri Aug 31 11:14:22 2012
From: govoiper at gmail.com (SamyGo)
Date: Fri, 31 Aug 2012 12:14:22 +0500
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <AD70ED57-CDBB-4F9D-AE0F-77008F9EB45E@me.com>
References: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
	<8D066FE8-DB24-4928-8508-317E23B858EA@me.com>
	<CAJUJwthxmhd0u6G35ANgiCzdHv1KY-qKK5ZD_Xx_+S-cGYMXVg@mail.gmail.com>
	<9065DE2C-DF4A-4E6C-B436-1A83B2AFA664@me.com>
	<CAJUJwthU62H9o3iutVpt3EZO2-UKW0PXk6fJgFeGUO7NT1W0VQ@mail.gmail.com>
	<AD70ED57-CDBB-4F9D-AE0F-77008F9EB45E@me.com>
Message-ID: <CAJUJwthhd8jj4ZGeFnFv-oRS8Xvst_T+a10_ecbomrwG58-LXw@mail.gmail.com>

Well that narrow downs the possible solutions to doing some stuff on FS
only ! - I really think that Stopping or applying a work-around for the
re-invite from Broadsoft will get this fixed. I did some googling on the
SIP 491 code and re-invites in FS and mostly they point to session timers
or some media_bridging option in dialplan.


On Fri, Aug 31, 2012 at 12:03 PM, Mike Burlingame <mike.burlingame at me.com>wrote:

> The Sonus and M6 are out of my control they belong to my upstream and
> downstream so I have no access to them
>
> Broadsoft support will not work this issue because FS ACKs the 200 OK so
> dialog is completed at least on the B-Leg so that's when the M6 sends the
> 2nd invite however the dialog has not completed on the A-Leg but FS still
> passes the Invite along resulting in the 491
>
>
> Sent from my iPhone 4S
>
> On Aug 30, 2012, at 11:42 PM, SamyGo <govoiper at gmail.com> wrote:
>
> OK, and can you see whats happening inside the SONUS when it gets the
> second invite (debug logs etc)? Can you control the M6 to stop re-invites !?
>
> On Fri, Aug 31, 2012 at 10:29 AM, Mike Burlingame <mike.burlingame at me.com>wrote:
>
>> Sure I will put logs and sip dump up tomorrow A-Leg == Sonus B-Leg ==
>> Broadsoft M6
>>
>> Sent from my iPhone 4S
>>
>> On Aug 30, 2012, at 10:24 PM, SamyGo <govoiper at gmail.com> wrote:
>>
>> Hi,
>> Can you please manage a proper sip trace for this ! I'd really love to
>> troubleshoot and analyse the sip traces.
>> What is the equipment on your A-leg ?
>>
>> BR
>> Sammy
>>
>>
>>
>> On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame at me.com>wrote:
>>
>>> I would like to add an update to this and maybe someone has some other
>>> suggestions - It does look like that FS is disconnecting the call with
>>> cause code 31  [NORMAL_UNSPECIFIED] I did not notice the time stamp
>>> difference on the Bye when I first looked at the log's
>>>
>>> The basis of the issue still stands a 491 Results in the B-Leg being
>>> disconnected
>>>
>>> On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame at me.com>
>>> wrote:
>>>
>>> We are seeing some instances when we send a invite from the B-Leg back
>>> to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg
>>> has not fully completed yet causing a return of a 491 from the A-Leg side
>>> causing the call to be disconnected. wanted to see if anyone else has seen
>>> something like this while running FS and if anyone had any suggestions on a
>>> fix?
>>>
>>> A-Leg
>>> Invite into Freeswitch
>>> 100 Trying back from FS to A-Leg
>>> 180 Ringing from FS to A-Leg
>>> 200 OK from FS to A-Leg at 15:08:14.638799
>>> Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515
>>> 100 Giving a try form A-Leg to FS at 15:08:14.749757
>>> 491 From A-Leg to FS at 15:08:14.780968
>>> ACK from FS to A-Leg at 15:08:14.781102
>>> ACK from FS to A-Leg at 15:08:14.797143
>>> BYE from A-LEG to FS 15:11:10.791963
>>>
>>>    481 Call Does Not Exist back to A-LEG
>>>
>>>
>>> B-Leg
>>> Invite from Freeswitch to B-Leg
>>> 100 Giving a try from B-Leg
>>> 180 Ringing from B-Leg
>>> 200 OK  from B-Leg at 15:08:14.635670
>>> ACK from FS to B-Leg  at 15:08:14.637044
>>> Invite from B-Leg to FS at 15:08:14.748623
>>> 100 Trying from FS to B-Leg at 15:08:14.748954
>>> 491 from FS to B-Leg at 15:08:14.782169
>>> ACK from B-Leg to FS at 15:08:14.782372
>>> BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 avi at avimarcus.net  Fri Aug 31 11:25:01 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 31 Aug 2012 10:25:01 +0300
Subject: [Freeswitch-users] No Dialplan on answer channel
In-Reply-To: <CAKzWOxXyYZH8UUSDEVzvcF+avzFLYQaT-6sNbvsozSiW-FmsJQ@mail.gmail.com>
References: <CAO-yT6hxe8eCcuT36WEjqmLDKYYfyGHnv3c2O_Opz3q6UMOhmA@mail.gmail.com>
	<CAKzWOxXyYZH8UUSDEVzvcF+avzFLYQaT-6sNbvsozSiW-FmsJQ@mail.gmail.com>
Message-ID: <CALNZuwnKiXKR6S4eqiuOizu=1C+rBuGpxPnrzHDnU3cu9RvB=w@mail.gmail.com>

The wiki tells you alternative syntaxes:
http://wiki.freeswitch.org/wiki/Mod_commands#originate

e.g. originate sofia/example/1000 at somewhere.com &javascript(test.js)

... but if the quotes doesn't work, the next wiki entry is wrong...

-Avi


On Fri, Aug 31, 2012 at 8:53 AM, Michael Collins <msc at freeswitch.org> wrote:

> It might be the location of your single quotes. Try something a little
> different, like this:
>
> *originate sofia/external/1000 at somewhere.com &javascript('hello.js
> /opt/freeswitch/sounds/2.mp3')*
>
> If that doesn't work then keep tinkering. If you still need more help then
> I recommend putting your console output into pastebin.freeswitch.org and
> linking back here in this thread. Be sure to use "FreeSWITCH Log" as the
> syntax highlighting.
>
> -MC
>
> On Thu, Aug 30, 2012 at 9:02 PM, Khue Nguyen Minh <khuenm at vega.com.vn>wrote:
>
>> Hi all,
>>
>> I want make outbound call to a user with command:
>>
>> *originate sofia/external/1000 at somewhere.com '&javascript(hello.js
>> /opt/freeswitch/sounds/2.mp3)'*
>>
>> my javascript will play file /opt/freeswitch/sounds/2.mp3.
>>
>> The call make successful but hangup immediately. I see log in fs_cli and
>> see this line "No Dialplan on answer channel". How I can fix this problem?
>>
>> Thanks
>> Khue Nguyen.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 avi at avimarcus.net  Fri Aug 31 11:44:38 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 31 Aug 2012 10:44:38 +0300
Subject: [Freeswitch-users] endpoint /user and sofia_contact change?
In-Reply-To: <CAKbxfG9QYWebJ0RkH21yYxD2sLjjhJ+-7PYNVLJ8aNe2zU6kFA@mail.gmail.com>
References: <CALNZuwnv86enuhdOm57zSwF7G-D9U4gFnUBZR8R+QWjC5Wu4zw@mail.gmail.com>
	<CAC-LwPMoGO2GCeMxC3Yw6-FV6wzwA6LvTyFKbmWt9O8s7Ua9pw@mail.gmail.com>
	<CALNZuwmmm_3yo2tdYT_58M7Qgbz7QiRzmf42kpCwta5psD4YFg@mail.gmail.com>
	<CAKbxfG9QYWebJ0RkH21yYxD2sLjjhJ+-7PYNVLJ8aNe2zU6kFA@mail.gmail.com>
Message-ID: <CALNZuwmsOvxTU3aEy=07_GNzU7_UtD+KNkPKwc70nz1q9YyqAg@mail.gmail.com>

Ah, ok. This needs to be noted somewhere....

Also, how does this affect bridging to "user", e.g. user/1000 ? I've never
seen the profile in one of those.

-Avi


On Fri, Aug 31, 2012 at 3:29 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Reverse that, a bug was fixed that may now appear as a bug, add the
> special profile name * in your sofia_contact call so it searches all
> profiles...
>
> sofia_contact */user at domain.com
> On Aug 29, 2012 10:36 AM, "Avi Marcus" <avi at avimarcus.net> wrote:
>
>> freeswitch at default> sofia_contact 1000
>> error/user_not_registered
>>
>> freeswitch at default> sofia_contact 1000 at sip.mydomain.com
>> error/user_not_registered
>>
>> freeswitch at default> sofia_contact internal/1000 at sip.mydomain.com
>> sofia/internal/sip:1000 at 192.168.255.103:5077,sofia/internal/
>> sip:1000 at 192.168.255.103:5078
>> ,sofia/internal/sip:1000 at 192.168.255.103:32998;ob
>>
>> The first 2 worked before. I suppose this is a bug.
>>
>> -Avi
>>
>>
>> On Wed, Aug 29, 2012 at 6:23 PM, Vik Killa <vipkilla at gmail.com> wrote:
>>
>>> does 'sofia status profile <profile_name> reg' show them as registered?
>>> try registering the handsets or flushing sofia registrations and
>>> re-registering phone.
>>>
>>> On Wed, Aug 29, 2012 at 11:07 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>> > I think I had 1.2 release yesterday and I would bridge to user/1000 or
>>> user
>>> > 1000 at mydomain.com and/or do sofia_contact(1000) and all is fine.
>>> > However, since building on master in GIT last night: FreeSWITCH Version
>>> > 1.3.0+git~20120828T230559Z~f1d201f03b (1.3.0; git at commit f1d201f03b
>>> on
>>> > Tue, 28 Aug 2012 23:05:59 Z)
>>> >
>>> > They return not registered. I have to do sofia_contact internal/1000.
>>> I'm
>>> > not sure how to use user/ I replaced it with
>>> > ${sofia_contact(internal/1000)}.
>>> > Was this an intentional change?
>>> > Is there some list of breaking changes somewhere?
>>> >
>>> > -Avi
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.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
>>> >
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 gavin.henry at gmail.com  Fri Aug 31 12:30:35 2012
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 31 Aug 2012 09:30:35 +0100
Subject: [Freeswitch-users] cisco spa 500 configs?
In-Reply-To: <CAKzWOxWc68uMp0R+NCVATm6YZfxT6DEn7v9whwoyMKBDzXWkmg@mail.gmail.com>
References: <CAKzWOxWc68uMp0R+NCVATm6YZfxT6DEn7v9whwoyMKBDzXWkmg@mail.gmail.com>
Message-ID: <CAA8_NKD5Oy+RP6mbPnJd1_=aymzRKvgfTx3hxxwjQ2rx_N_1mA@mail.gmail.com>

On 31 August 2012 02:07, Michael Collins <msc at freeswitch.org> wrote:
> Does anyone have a working config for a spa 5xx phone w/ s500 sidecar? I'd
> like to get that and the SPA 300 phones better documented on the wiki.

We have this, but haven't added S500DS/S details yet:

http://www.surevoip.co.uk/support/wiki/howtos:setup:cisco

Thanks.

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


From support at sping.nl  Fri Aug 31 11:13:50 2012
From: support at sping.nl (Systeembeheer)
Date: Fri, 31 Aug 2012 09:13:50 +0200
Subject: [Freeswitch-users] Pause in bridge
Message-ID: <5040642E.90607@sping.nl>

Hi,

I'm trying to configure the following: When a call comes in, it is first 
bridged to one extension and after 10 seconds to a second extension.

It is possible to issue the bridge command with a timeout and continue 
settings to one extension and a second bridge command to both 
extensions. There's some problems with that:
- The first extension can answer and 'just miss' the first bridge attempt.
- If the first call is cancelled by the user, it rings again 10 sec. later.

Is there a way to bridge to an extension and bridge to a second 
extension 10 sec. later without breaking the first bridge? Or bridge to 
extensions with a pause of 10 sec. for one of them?


Thanks!

steveN


From avi at avimarcus.net  Fri Aug 31 13:07:20 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 31 Aug 2012 12:07:20 +0300
Subject: [Freeswitch-users] Pause in bridge
In-Reply-To: <5040642E.90607@sping.nl>
References: <5040642E.90607@sping.nl>
Message-ID: <CALNZuwmMdodWpCDsDZ2pLuVOpudP1PEGFo=XQyPigSeLVUA6wA@mail.gmail.com>

See http://wiki.freeswitch.org/wiki/Variable_leg_delay_start
-Avi


On Fri, Aug 31, 2012 at 10:13 AM, Systeembeheer <support at sping.nl> wrote:

> Hi,
>
> I'm trying to configure the following: When a call comes in, it is first
> bridged to one extension and after 10 seconds to a second extension.
>
> It is possible to issue the bridge command with a timeout and continue
> settings to one extension and a second bridge command to both
> extensions. There's some problems with that:
> - The first extension can answer and 'just miss' the first bridge attempt.
> - If the first call is cancelled by the user, it rings again 10 sec. later.
>
> Is there a way to bridge to an extension and bridge to a second
> extension 10 sec. later without breaking the first bridge? Or bridge to
> extensions with a pause of 10 sec. for one of them?
>
>
> Thanks!
>
> steveN
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-list at puzzled.xs4all.nl  Fri Aug 31 14:13:54 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 31 Aug 2012 12:13:54 +0200
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <CAPcNkddX0xXdJsYm+UTwy=xo5Saw5gxk=5XF_rMcJi7yTZ+mzQ@mail.gmail.com>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
	<CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
	<CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
	<CAA8_NKAnX97E0dV7tpakg+wO-TzWdKm5wrm0qNP1EqODAVRoNw@mail.gmail.com>
	<CAPNTnnpHzcUzJ8Jnkj6Nma_2pZVyVhfWDA4RjBtJkuNWEMOdvg@mail.gmail.com>
	<0D1C698866F66045A6201FD0F59CAC90014678ACA7@EX.frontier.local>
	<63B00DD1DA6A364E9F64A3A0BD2FE7B612FC3248@BLUPRD0711MB413.namprd07.prod.outlook.com>
	<CAPcNkddX0xXdJsYm+UTwy=xo5Saw5gxk=5XF_rMcJi7yTZ+mzQ@mail.gmail.com>
Message-ID: <50408E62.7070400@puzzled.xs4all.nl>

On 31-08-12 05:45, Ken Rice wrote:
> the problem happens at somewhere around 50 to 100 CPS, system % goes
> thru the roof, and its not IO.. loglevel 0... etc... now its also worth
> mentioning that on 6.2 the number of context switches for similar
> amounts of calls on the same physical hardware seems to be double vs
> something like cent5 or deb6....

Is this also the case for 6.3 with all post-6.3 updates applied?

Regards,
Patrick



From ntomer at newgen.co.in  Fri Aug 31 14:50:12 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Fri, 31 Aug 2012 16:20:12 +0530
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded to
	caller	itself
In-Reply-To: <018601cd85ce$51c89730$f559c590$@co.in>
References: <018601cd85ce$51c89730$f559c590$@co.in>
Message-ID: <016901cd8766$689f0b80$39dd2280$@co.in>

Any idea why it might be happening?

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin
Tomer
Sent: Wednesday, August 29, 2012 3:39 PM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded to caller
itself

 

Hi,

 

I've configure mod_callcenter on Ubuntu 12.0.4. I've 5 extensions working on
6 soft phones - 1000 to 1005. 1000 is used to call the IVR number, rest 5
are agents.

 

The dial plan looks like -

 

<extension name="ivr_nitin">

      <condition field="destination_number" expression="^5002$">

        <action application="answer"/>

        <action application="sleep" data="500"/>

                <action application="ivr" data="nitin_ivr"/>

      </condition>

    </extension>

<!-- For Call Center -->

<!-- Newgen Support -->

       <extension>

          <condition field="destination_number" expression="^(450)$">

              <action application="set" data="caller_id_name=Newgen Support"
/>

              <action application="set" data="call_timeout=60" />

              <action application="set" data="originate_timeout=60" />

              <action application="callcenter" data="support at default"/>

           </condition>

        </extension>

 

<!-- Newgen Sales -->

        <extension>

           <condition field="destination_number" expression="^(451)$">

               <action application="set" data="caller_id_name=Newgen Sales"
/>

               <action application="set" data="call_timeout=60" />

               <action application="set" data="originate_timeout=60" />

               <action application="callcenter" data="sales at default"/>

           </condition>

        </extension>

 

<!-- Newgen Human Resources -->

        <extension>

           <condition field="destination_number" expression="^(452)$">

               <action application="set" data="caller_id_name=Newgen Human
Resources" />

                       <action application="set" data="call_timeout=60" />

               <action application="set" data="originate_timeout=60" />

               <action application="callcenter" data="hr at default"/>

           </condition>

        </extension>

 

IVR.xml -

 

<include>

        <menu name="nitin_ivr"

                greet-long="say:Welcome to Newgen Software Technologies
Limited. Press 1 for support, 2 for sales or 3 for human resource"

                greet-short="say:Welcome to Newgen. Press 1 for support, 2
for sales or 3 for human resources"

                invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"

                exit-sound="voicemail/vm-goodbye.wav"

                confirm-macro=""

                confirm-key=""

                tts-engine="flite"

                tts-voice="slt"

                confirm-attempts="3"

                timeout="3000"

                inter-digit-timeout="2000"

                max-failures="3"

                max-timeouts="3"

                digit-len="4">

 

                <entry action="menu-exec-app" digits="1" param="transfer 450
XML default"/>    <!-- Support -->

                <entry action="menu-exec-app" digits="2" param="transfer 451
XML default"/>    <!-- Sales -->

                <entry action="menu-exec-app" digits="3" param="transfer 452
XML default"/>    <!-- Human Resources -->

 

        </menu>

</include>

 

And callcenter.conf.xml -

 

<configuration name="callcenter.conf" description="CallCenter">

  <settings>

    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->

    <!--<param name="dbname" value="/dev/shm/callcenter.db"/>-->

  </settings>

 

    <queues>

                <queue name="sales at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/sales/${strftime(%Y-%m-%d-%H-%M-%S)}.${destin
ation_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="14400"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

                <queue name="support at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${dest
ination_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="60"/>

                                <param name="abandoned-resume-allowed"
value="false"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

                <queue name="hr at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${dest
ination_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="60"/>

                                <param name="abandoned-resume-allowed"
value="false"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

  </queues>

 

<!-- WARNING: Configuration of XML Agents will be updated into the DB upon
restart. -->

<!-- WARNING: Configuration of XML Tiers will reset the level and position
if those were supplied. -->

<!-- WARNING: Agents and Tiers XML config shouldn't be used in a multi FS
shared DB setup (Not currently supported anyway) -->

  <agents>

    <agent name="1001" type="callback" contact="[call_timeout=10]user/1001"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1002" type="callback" contact="[call_timeout=10]user/1002"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1003" type="callback" contact="[call_timeout=10]user/1003"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1004" type="callback" contact="[call_timeout=10]user/1004"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1005" type="callback" contact="[call_timeout=10]user/1005"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

  </agents>

  <tiers>

    <!-- If no level or position is provided, they will default to 1.  You
should do this to keep db value on restart. -->

                <!-- agent 1001 is in all three queues -->

                <tier agent="1001" queue="sales at default" level="1"
position="1"/>

                <tier agent="1001" queue="support at default" level="1"
position="1"/>

                <tier agent="1001" queue="hr at default" level="1"
position="1"/>

 

                <tier agent="1002" queue="sales at default" level="1"
position="1"/>

 

                <tier agent="1003" queue="support at default" level="1"
position="1"/>

 

                <tier agent="1004" queue="sales at default" level="1"
position="1"/>

                <tier agent="1004" queue="support at default" level="1"
position="1"/>

 

                <tier agent="1005" queue="hr at default" level="1"
position="1"/>

  </tiers>

 

</configuration>

 

When I call from extension 1000 and press 2 for sales, call is routed back
to extension 1000, which is not even in agents' list. Why is it happening?

 

Regards

 

Nitin

 

 


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From vbvbrj at gmail.com  Fri Aug 31 16:14:50 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Fri, 31 Aug 2012 15:14:50 +0300
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded to
 caller itself
In-Reply-To: <016901cd8766$689f0b80$39dd2280$@co.in>
References: <018601cd85ce$51c89730$f559c590$@co.in>
	<016901cd8766$689f0b80$39dd2280$@co.in>
Message-ID: <5040AABA.8090707@gmail.com>

On 31.08.2012 13:50, Nitin Tomer wrote:
> Any idea why it might be happening?

Pastebin the log in debug mode. Also, analyze your dialplan, maybe there 
is some other extension interfering.

-- 
Mimiko desu.


From jayesh.voip at gmail.com  Fri Aug 31 17:25:21 2012
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Fri, 31 Aug 2012 18:55:21 +0530
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded to
	caller itself
Message-ID: <CALvF6vCkHzwRFyP+RfRMvu-9Yfh8iWQ2+yhzYDhc_30_wq-4+w@mail.gmail.com>

Nitin,

> If this is   your exact configuration, it is simply not possible that the
> call gets routed back to same extension unless there is something
> configured in the dialplan.

Look at the console in loglevel 7 mode and check if the call goes into
callcenter module or no? If no, check the flow of your dialplan and figure
out what is taking the call back to the calling extension.

--- Jayesh


> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nitin Tomer
>
> *Sent:* Wednesday, August 29, 2012 3:39 PM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* [Freeswitch-users] mod_callcenter: Call being forwarded to
> caller itself****
>
> ** **
>
> Hi,****
>
> ** **
>
> I?ve configure mod_callcenter on Ubuntu 12.0.4. I?ve 5 extensions working
> on 6 soft phones ? 1000 to 1005. 1000 is used to call the IVR number, rest
> 5 are agents.****
>
> ** **
>
> The dial plan looks like ?****
>
> ** **
>
> <extension name="ivr_nitin">****
>
>       <condition field="destination_number" expression="^5002$">****
>
>         <action application="answer"/>****
>
>         <action application="sleep" data="500"/>****
>
>                 <action application="ivr" data="nitin_ivr"/>****
>
>       </condition>****
>
>     </extension>****
>
> <!-- For Call Center -->****
>
> <!-- Newgen Support -->****
>
>        <extension>****
>
>           <condition field="destination_number" expression="^(450)$">****
>
>               <action application="set" data="caller_id_name=Newgen
> Support" />****
>
>               <action application="set" data="call_timeout=60" />****
>
>               <action application="set" data="originate_timeout=60" />****
>
>               <action application="callcenter" data="support at default"/>***
> *
>
>            </condition>****
>
>         </extension>****
>
>  ****
>
> <!-- Newgen Sales -->****
>
>         <extension>****
>
>            <condition field="destination_number" expression="^(451)$">****
>
>                <action application="set" data="caller_id_name=Newgen
> Sales" />****
>
>                <action application="set" data="call_timeout=60" />****
>
>                <action application="set" data="originate_timeout=60" />***
> *
>
>                <action application="callcenter" data="sales at default"/>****
>
>            </condition>****
>
>         </extension>****
>
> ** **
>
> <!-- Newgen Human Resources -->****
>
>         <extension>****
>
>            <condition field="destination_number" expression="^(452)$">****
>
>                <action application="set" data="caller_id_name=Newgen Human
> Resources" />****
>
>                        <action application="set" data="call_timeout=60" />
> ****
>
>                <action application="set" data="originate_timeout=60" />***
> *
>
>                <action application="callcenter" data="hr at default"/>****
>
>            </condition>****
>
>         </extension>****
>
> ** **
>
> IVR.xml ?****
>
> ** **
>
> <include>****
>
>         <menu name="nitin_ivr"****
>
>                 greet-long="say:Welcome to Newgen Software Technologies
> Limited. Press 1 for support, 2 for sales or 3 for human resource"****
>
>                 greet-short="say:Welcome to Newgen. Press 1 for support, 2
> for sales or 3 for human resources"****
>
>                 invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"****
>
>                 exit-sound="voicemail/vm-goodbye.wav"****
>
>                 confirm-macro=""****
>
>                 confirm-key=""****
>
>                 tts-engine="flite"****
>
>                 tts-voice="slt"****
>
>                 confirm-attempts="3"****
>
>                 timeout="3000"****
>
>                 inter-digit-timeout="2000"****
>
>                 max-failures="3"****
>
>                 max-timeouts="3"****
>
>                 digit-len="4">****
>
>  ****
>
>                 <entry action="menu-exec-app" digits="1" param="transfer
> 450 XML default"/>    <!-- Support -->****
>
>                 <entry action="menu-exec-app" digits="2" param="transfer
> 451 XML default"/>    <!-- Sales -->****
>
>                 <entry action="menu-exec-app" digits="3" param="transfer
> 452 XML default"/>    <!-- Human Resources -->****
>
> ** **
>
>         </menu>****
>
> </include>****
>
> ** **
>
> And callcenter.conf.xml ?****
>
> ** **
>
> <configuration name="callcenter.conf" description="CallCenter">****
>
>   <settings>****
>
>     <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->****
>
>     <!--<param name="dbname" value="/dev/shm/callcenter.db"/>-->****
>
>   </settings>****
>
> ** **
>
>     <queues>****
>
>                 <queue name="sales at default">****
>
>                                 <param name="strategy"
> value="agent-with-least-talk-time"/>****
>
>                                 <param name="moh-sound"
> value="$${hold_music}"/>****
>
>       <!--<param name="record-template"
> value="$${base_dir}/recordings/sales/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>-->
> ****
>
>                                 <param name="time-base-score"
> value="queue"/>****
>
>                                 <param name="tier-rules-apply"
> value="false"/>****
>
>                                 <param name="tier-rule-wait-second"
> value="300"/>****
>
>                                 <param
> name="tier-rule-wait-multiply-level" value="true"/>****
>
>                                 <param name="tier-rule-no-agent-no-wait"
> value="false"/>****
>
>                                 <param name="discard-abandoned-after"
> value="14400"/>****
>
>                                 <param name="max-wait-time" value="0"/>***
> *
>
>                                 <param name="max-wait-time-with-no-agent"
> value="120"/>****
>
>                 </queue>****
>
>                 <queue name="support at default">****
>
>                                 <param name="strategy"
> value="agent-with-least-talk-time"/>****
>
>                                 <param name="moh-sound"
> value="$${hold_music}"/>****
>
>       <!--<param name="record-template"
> value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>-->
> ****
>
>                                 <param name="time-base-score"
> value="queue"/>****
>
>                                 <param name="tier-rules-apply"
> value="false"/>****
>
>                                 <param name="tier-rule-wait-second"
> value="300"/>****
>
>                                 <param
> name="tier-rule-wait-multiply-level" value="true"/>****
>
>                                 <param name="tier-rule-no-agent-no-wait"
> value="false"/>****
>
>                                 <param name="discard-abandoned-after"
> value="60"/>****
>
>                                 <param name="abandoned-resume-allowed"
> value="false"/>****
>
>                                 <param name="max-wait-time" value="0"/>***
> *
>
>                                 <param name="max-wait-time-with-no-agent"
> value="120"/>****
>
>                 </queue>****
>
>                 <queue name="hr at default">****
>
>                                 <param name="strategy"
> value="agent-with-least-talk-time"/>****
>
>                                 <param name="moh-sound"
> value="$${hold_music}"/>****
>
>       <!--<param name="record-template"
> value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>-->
> ****
>
>                                 <param name="time-base-score"
> value="queue"/>****
>
>                                 <param name="tier-rules-apply"
> value="false"/>****
>
>                                 <param name="tier-rule-wait-second"
> value="300"/>****
>
>                                 <param
> name="tier-rule-wait-multiply-level" value="true"/>****
>
>                                 <param name="tier-rule-no-agent-no-wait"
> value="false"/>****
>
>                                 <param name="discard-abandoned-after"
> value="60"/>****
>
>                                 <param name="abandoned-resume-allowed"
> value="false"/>****
>
>                                 <param name="max-wait-time" value="0"/>***
> *
>
>                                 <param name="max-wait-time-with-no-agent"
> value="120"/>****
>
>                 </queue>****
>
>   </queues>****
>
> ** **
>
> <!-- WARNING: Configuration of XML Agents will be updated into the DB upon
> restart. -->****
>
> <!-- WARNING: Configuration of XML Tiers will reset the level and position
> if those were supplied. -->****
>
> <!-- WARNING: Agents and Tiers XML config shouldn't be used in a multi FS
> shared DB setup (Not currently supported anyway) -->****
>
>   <agents>****
>
>     <agent name="1001" type="callback"
> contact="[call_timeout=10]user/1001" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>      <agent name="1002" type="callback"
> contact="[call_timeout=10]user/1002" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>      <agent name="1003" type="callback"
> contact="[call_timeout=10]user/1003" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>      <agent name="1004" type="callback"
> contact="[call_timeout=10]user/1004" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>      <agent name="1005" type="callback"
> contact="[call_timeout=10]user/1005" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>   </agents>****
>
>   <tiers>****
>
>     <!-- If no level or position is provided, they will default to 1.  You
> should do this to keep db value on restart. -->****
>
>                 <!-- agent 1001 is in all three queues -->****
>
>                 <tier agent="1001" queue="sales at default" level="1"
> position="1"/>****
>
>                 <tier agent="1001" queue="support at default" level="1"
> position="1"/>****
>
>                 <tier agent="1001" queue="hr at default" level="1"
> position="1"/>****
>
> ** **
>
>                 <tier agent="1002" queue="sales at default" level="1"
> position="1"/>****
>
> ** **
>
>                 <tier agent="1003" queue="support at default" level="1"
> position="1"/>****
>
> ** **
>
>                 <tier agent="1004" queue="sales at default" level="1"
> position="1"/>****
>
>                 <tier agent="1004" queue="support at default" level="1"
> position="1"/>****
>
> ** **
>
>                 <tier agent="1005" queue="hr at default" level="1"
> position="1"/>****
>
>   </tiers>****
>
> ** **
>
> </configuration>****
>
> ** **
>
> When I call from extension 1000 and press 2 for sales, call is routed back
> to extension 1000, which is not even in agents? list. Why is it happening?
> ****
>
> ** **
>
> Regards****
>
> ** **
>
> Nitin****
>
>
> **
>
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From jayesh.voip at gmail.com  Fri Aug 31 17:29:36 2012
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Fri, 31 Aug 2012 18:59:36 +0530
Subject: [Freeswitch-users] mod_callcenter record template
Message-ID: <CALvF6vBPbi+rmnHPHvYVuUnMVyMOJmQLaNB29jO7AZzFRmaUMw@mail.gmail.com>

Hi,
Could you figure out a solution to this. The variable is ${cc_agent}.

--- Jayesh


> In mod callcenter.conf.xml there is:
> <param name="record-template" value="$${base_dir}/**
> recordings/sales/${strftime(%**Y-%m-%d-%H-%M-%S)}.${**
> destination_number}.${caller_**id_number}.${uuid}.wav"/>
> This names the file with callers number and callcenter's assigned number.
> Bit how to include in the file name the agent's number, which responded?
> Which variable to use?
>
>
>
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From krice at freeswitch.org  Fri Aug 31 17:34:40 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 31 Aug 2012 08:34:40 -0500
Subject: [Freeswitch-users] RedHat 6.X Performance
In-Reply-To: <50408E62.7070400@puzzled.xs4all.nl>
References: <CAK9Kg4Hb04R0zHcOQN7E_ySWUsA4WLip7aieprxvhSWRdAdZcw@mail.gmail.com>
	<CAPcNkdeA4FP_CaqFJtCm-4_7-o3J7nb-qUT8wPRCX6BeCOkAPQ@mail.gmail.com>
	<CAC-LwPPyPnjhL+S_8aA7_7OYFX_9N0m7WcUf3+sHwm-iuUmh9Q@mail.gmail.com>
	<CAKzWOxUDHyn8HzssgEGN56HqkEKzFZAzZyeUVvLexEyUdJwipg@mail.gmail.com>
	<CAPcNkdfDCDqBzsZ-Sgq+7548wRwd7BmM_t+CHjbhFPE5_2sj4A@mail.gmail.com>
	<CAA8_NKAnX97E0dV7tpakg+wO-TzWdKm5wrm0qNP1EqODAVRoNw@mail.gmail.com>
	<CAPNTnnpHzcUzJ8Jnkj6Nma_2pZVyVhfWDA4RjBtJkuNWEMOdvg@mail.gmail.com>
	<0D1C698866F66045A6201FD0F59CAC90014678ACA7@EX.frontier.local>
	<63B00DD1DA6A364E9F64A3A0BD2FE7B612FC3248@BLUPRD0711MB413.namprd07.prod.outlook.com>
	<CAPcNkddX0xXdJsYm+UTwy=xo5Saw5gxk=5XF_rMcJi7yTZ+mzQ@mail.gmail.com>
	<50408E62.7070400@puzzled.xs4all.nl>
Message-ID: <CAPcNkdd22hZ9NQ8TacvVJJL0=CP9qN9O89F=Nk9HYZyHbqLKbw@mail.gmail.com>

this was the case with all the updates applied at the last time i tried it
ie: yum update -y



On Fri, Aug 31, 2012 at 5:13 AM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 31-08-12 05:45, Ken Rice wrote:
> > the problem happens at somewhere around 50 to 100 CPS, system % goes
> > thru the roof, and its not IO.. loglevel 0... etc... now its also worth
> > mentioning that on 6.2 the number of context switches for similar
> > amounts of calls on the same physical hardware seems to be double vs
> > something like cent5 or deb6....
>
> Is this also the case for 6.3 with all post-6.3 updates applied?
>
> Regards,
> Patrick
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ntomer at newgen.co.in  Fri Aug 31 17:53:20 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Fri, 31 Aug 2012 19:23:20 +0530
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded
	to	caller itself
In-Reply-To: <CALvF6vCkHzwRFyP+RfRMvu-9Yfh8iWQ2+yhzYDhc_30_wq-4+w@mail.gmail.com>
References: <CALvF6vCkHzwRFyP+RfRMvu-9Yfh8iWQ2+yhzYDhc_30_wq-4+w@mail.gmail.com>
Message-ID: <019001cd877f$fb93dba0$f2bb92e0$@co.in>

Hi Jayesh,

 

Call is being redirected to call center because it is working other two
options. Even for "sales" the call is redirected to caller(extn 1000) but
when it is declined from there, it works as desired.

 

Thanks

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jayesh
Nambiar
Sent: Friday, August 31, 2012 6:55 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_callcenter: Call being forwarded to
caller itself

 

Nitin, 

If this is   your exact configuration, it is simply not possible that the
call gets routed back to same extension unless there is something configured
in the dialplan.

Look at the console in loglevel 7 mode and check if the call goes into
callcenter module or no? If no, check the flow of your dialplan and figure
out what is taking the call back to the calling extension.

 

--- Jayesh 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin
Tomer

Sent: Wednesday, August 29, 2012 3:39 PM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded to caller
itself

 

Hi,

 

I've configure mod_callcenter on Ubuntu 12.0.4. I've 5 extensions working on
6 soft phones - 1000 to 1005. 1000 is used to call the IVR number, rest 5
are agents.

 

The dial plan looks like -

 

<extension name="ivr_nitin">

      <condition field="destination_number" expression="^5002$">

        <action application="answer"/>

        <action application="sleep" data="500"/>

                <action application="ivr" data="nitin_ivr"/>

      </condition>

    </extension>

<!-- For Call Center -->

<!-- Newgen Support -->

       <extension>

          <condition field="destination_number" expression="^(450)$">

              <action application="set" data="caller_id_name=Newgen Support"
/>

              <action application="set" data="call_timeout=60" />

              <action application="set" data="originate_timeout=60" />

              <action application="callcenter" data="support at default"/>

           </condition>

        </extension>

 

<!-- Newgen Sales -->

        <extension>

           <condition field="destination_number" expression="^(451)$">

               <action application="set" data="caller_id_name=Newgen Sales"
/>

               <action application="set" data="call_timeout=60" />

               <action application="set" data="originate_timeout=60" />

               <action application="callcenter" data="sales at default"/>

           </condition>

        </extension>

 

<!-- Newgen Human Resources -->

        <extension>

           <condition field="destination_number" expression="^(452)$">

               <action application="set" data="caller_id_name=Newgen Human
Resources" />

                       <action application="set" data="call_timeout=60" />

               <action application="set" data="originate_timeout=60" />

               <action application="callcenter" data="hr at default"/>

           </condition>

        </extension>

 

IVR.xml -

 

<include>

        <menu name="nitin_ivr"

                greet-long="say:Welcome to Newgen Software Technologies
Limited. Press 1 for support, 2 for sales or 3 for human resource"

                greet-short="say:Welcome to Newgen. Press 1 for support, 2
for sales or 3 for human resources"

                invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"

                exit-sound="voicemail/vm-goodbye.wav"

                confirm-macro=""

                confirm-key=""

                tts-engine="flite"

                tts-voice="slt"

                confirm-attempts="3"

                timeout="3000"

                inter-digit-timeout="2000"

                max-failures="3"

                max-timeouts="3"

                digit-len="4">

 

                <entry action="menu-exec-app" digits="1" param="transfer 450
XML default"/>    <!-- Support -->

                <entry action="menu-exec-app" digits="2" param="transfer 451
XML default"/>    <!-- Sales -->

                <entry action="menu-exec-app" digits="3" param="transfer 452
XML default"/>    <!-- Human Resources -->

 

        </menu>

</include>

 

And callcenter.conf.xml -

 

<configuration name="callcenter.conf" description="CallCenter">

  <settings>

    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->

    <!--<param name="dbname" value="/dev/shm/callcenter.db"/>-->

  </settings>

 

    <queues>

                <queue name="sales at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/sales/${strftime(%Y-%m-%d-%H-%M-%S)}.${destin
ation_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="14400"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

                <queue name="support at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${dest
ination_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="60"/>

                                <param name="abandoned-resume-allowed"
value="false"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

                <queue name="hr at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${dest
ination_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="60"/>

                                <param name="abandoned-resume-allowed"
value="false"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

  </queues>

 

<!-- WARNING: Configuration of XML Agents will be updated into the DB upon
restart. -->

<!-- WARNING: Configuration of XML Tiers will reset the level and position
if those were supplied. -->

<!-- WARNING: Agents and Tiers XML config shouldn't be used in a multi FS
shared DB setup (Not currently supported anyway) -->

  <agents>

    <agent name="1001" type="callback" contact="[call_timeout=10]user/1001"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1002" type="callback" contact="[call_timeout=10]user/1002"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1003" type="callback" contact="[call_timeout=10]user/1003"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1004" type="callback" contact="[call_timeout=10]user/1004"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1005" type="callback" contact="[call_timeout=10]user/1005"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

  </agents>

  <tiers>

    <!-- If no level or position is provided, they will default to 1.  You
should do this to keep db value on restart. -->

                <!-- agent 1001 is in all three queues -->

                <tier agent="1001" queue="sales at default" level="1"
position="1"/>

                <tier agent="1001" queue="support at default" level="1"
position="1"/>

                <tier agent="1001" queue="hr at default" level="1"
position="1"/>

 

                <tier agent="1002" queue="sales at default" level="1"
position="1"/>

 

                <tier agent="1003" queue="support at default" level="1"
position="1"/>

 

                <tier agent="1004" queue="sales at default" level="1"
position="1"/>

                <tier agent="1004" queue="support at default" level="1"
position="1"/>

 

                <tier agent="1005" queue="hr at default" level="1"
position="1"/>

  </tiers>

 

</configuration>

 

When I call from extension 1000 and press 2 for sales, call is routed back
to extension 1000, which is not even in agents' list. Why is it happening?

 

Regards

 

Nitin

 


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From jayesh.voip at gmail.com  Fri Aug 31 18:07:10 2012
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Fri, 31 Aug 2012 19:37:10 +0530
Subject: [Freeswitch-users] Recording problem
Message-ID: <CALvF6vCS8tW_4_4ecGhrcU6ia+8A0BoAHSfquCLK2VnFpyHx4Q@mail.gmail.com>

Hi,
I had this same problem and I don't think it is related to mod_callcenter.
I had also sent a post in regards to this here:
http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/085552.html

I figured that this generally happened when my caller leg was connected on
G711ulaw and agent leg was connected on G722 codec. When both the caller
leg and agent leg was same the recordings were clear. Can someone please
confirm if this could be the problem !!

--- Jayesh


>    Sounds like network connection. Can you try switching stations between a
>>
>> bad phone and a good phone. Maybe your network has bad ports or cables
>> somewhere. I know for all cisco voip they recommend cat 6 cables.
>>
>
> It is definatelly not a network problem. The clients are connected to the
> same switch. I posted the log to see the difference. Observe "codec
> substitution" lines for bad clients. I don't know what this means to the
> system as why a substitution is done, but are not tried other codecs from
> client to compare.
>
>
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From jayesh.voip at gmail.com  Fri Aug 31 18:28:41 2012
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Fri, 31 Aug 2012 19:58:41 +0530
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded to
	caller itself
Message-ID: <CALvF6vA-Yma2ie7CBOrzpEw93K0s63p=eF+Nj6=SSbfp8a3DTw@mail.gmail.com>

Nitin,
Are you sure that the dialplan you pasted earlier is within the default
context? Since you are taking the logic to default context from the IVR.

--- Jayesh

>
> Pastebin the log in debug mode. Also, analyze your dialplan, maybe there
> is some other extension interfering.
>
> --
> Mimiko desu.
>
>
>
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From jayesh.voip at gmail.com  Fri Aug 31 18:36:36 2012
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Fri, 31 Aug 2012 20:06:36 +0530
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded to
	caller itself
Message-ID: <CALvF6vBeLSrW70vsPeAK0V4h-28Cr1y+_fD0wmx-0TLr1hTVrQ@mail.gmail.com>

Hi Nitin,
In that case all you have to confirm is that you have not set the contact
of any agent as 1000. Keep in mind that the agents can be pushed into
freeswitch memory from API commands also. Basically you can use FS_CLI to
add agents and set their contacts even without mentioning those agents in
the XML file. So just be sure that you do not have any agent's contact set
to 1000 and also check from FS_CLI issuing this command: "callcenter_config
agent list" to make sure that none of the agent has contact set to 1000.

--- Jayesh



>  Hi Jayesh,****
>
> ** **
>
> Call is being redirected to call center because it is working other two
> options. Even for ?sales? the call is redirected to caller(extn 1000) but
> when it is declined from there, it works as desired.****
>
> ** **
>
> Thanks****
>
> ** **
>
> Nitin****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayesh
> Nambiar
> *Sent:* Friday, August 31, 2012 6:55 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] mod_callcenter: Call being forwarded to
> caller itself****
>
> ** **
>
> Nitin, ****
>
> If this is   your exact configuration, it is simply not possible that the
> call gets routed back to same extension unless there is something
> configured in the dialplan.****
>
>  Look at the console in loglevel 7 mode and check if the call goes into
> callcenter module or no? If no, check the flow of your dialplan and figure
> out what is taking the call back to the calling extension.****
>
> ** **
>
> --- Jayesh ****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nitin Tomer*
> ***
>
> *Sent:* Wednesday, August 29, 2012 3:39 PM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* [Freeswitch-users] mod_callcenter: Call being forwarded to
> caller itself****
>
>  ****
>
> Hi,****
>
>  ****
>
> I?ve configure mod_callcenter on Ubuntu 12.0.4. I?ve 5 extensions working
> on 6 soft phones ? 1000 to 1005. 1000 is used to call the IVR number, rest
> 5 are agents.****
>
>  ****
>
> The dial plan looks like ?****
>
>  ****
>
> <extension name="ivr_nitin">****
>
>       <condition field="destination_number" expression="^5002$">****
>
>         <action application="answer"/>****
>
>         <action application="sleep" data="500"/>****
>
>                 <action application="ivr" data="nitin_ivr"/>****
>
>       </condition>****
>
>     </extension>****
>
> <!-- For Call Center -->****
>
> <!-- Newgen Support -->****
>
>        <extension>****
>
>           <condition field="destination_number" expression="^(450)$">****
>
>               <action application="set" data="caller_id_name=Newgen
> Support" />****
>
>               <action application="set" data="call_timeout=60" />****
>
>               <action application="set" data="originate_timeout=60" />****
>
>               <action application="callcenter" data="support at default"/>***
> *
>
>            </condition>****
>
>         </extension>****
>
>  ****
>
> <!-- Newgen Sales -->****
>
>         <extension>****
>
>            <condition field="destination_number" expression="^(451)$">****
>
>                <action application="set" data="caller_id_name=Newgen
> Sales" />****
>
>                <action application="set" data="call_timeout=60" />****
>
>                <action application="set" data="originate_timeout=60" />***
> *
>
>                <action application="callcenter" data="sales at default"/>****
>
>            </condition>****
>
>         </extension>****
>
>  ****
>
> <!-- Newgen Human Resources -->****
>
>         <extension>****
>
>            <condition field="destination_number" expression="^(452)$">****
>
>                <action application="set" data="caller_id_name=Newgen Human
> Resources" />****
>
>                        <action application="set" data="call_timeout=60" />
> ****
>
>                <action application="set" data="originate_timeout=60" />***
> *
>
>                <action application="callcenter" data="hr at default"/>****
>
>            </condition>****
>
>         </extension>****
>
>  ****
>
> IVR.xml ?****
>
>  ****
>
> <include>****
>
>         <menu name="nitin_ivr"****
>
>                 greet-long="say:Welcome to Newgen Software Technologies
> Limited. Press 1 for support, 2 for sales or 3 for human resource"****
>
>                 greet-short="say:Welcome to Newgen. Press 1 for support, 2
> for sales or 3 for human resources"****
>
>                 invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"****
>
>                 exit-sound="voicemail/vm-goodbye.wav"****
>
>                 confirm-macro=""****
>
>                 confirm-key=""****
>
>                 tts-engine="flite"****
>
>                 tts-voice="slt"****
>
>                 confirm-attempts="3"****
>
>                 timeout="3000"****
>
>                 inter-digit-timeout="2000"****
>
>                 max-failures="3"****
>
>                 max-timeouts="3"****
>
>                 digit-len="4">****
>
>  ****
>
>                 <entry action="menu-exec-app" digits="1" param="transfer
> 450 XML default"/>    <!-- Support -->****
>
>                 <entry action="menu-exec-app" digits="2" param="transfer
> 451 XML default"/>    <!-- Sales -->****
>
>                 <entry action="menu-exec-app" digits="3" param="transfer
> 452 XML default"/>    <!-- Human Resources -->****
>
>  ****
>
>         </menu>****
>
> </include>****
>
>  ****
>
> And callcenter.conf.xml ?****
>
>  ****
>
> <configuration name="callcenter.conf" description="CallCenter">****
>
>   <settings>****
>
>     <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->****
>
>     <!--<param name="dbname" value="/dev/shm/callcenter.db"/>-->****
>
>   </settings>****
>
>  ****
>
>     <queues>****
>
>                 <queue name="sales at default">****
>
>                                 <param name="strategy"
> value="agent-with-least-talk-time"/>****
>
>                                 <param name="moh-sound"
> value="$${hold_music}"/>****
>
>       <!--<param name="record-template"
> value="$${base_dir}/recordings/sales/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>-->
> ****
>
>                                 <param name="time-base-score"
> value="queue"/>****
>
>                                 <param name="tier-rules-apply"
> value="false"/>****
>
>                                 <param name="tier-rule-wait-second"
> value="300"/>****
>
>                                 <param
> name="tier-rule-wait-multiply-level" value="true"/>****
>
>                                 <param name="tier-rule-no-agent-no-wait"
> value="false"/>****
>
>                                 <param name="discard-abandoned-after"
> value="14400"/>****
>
>                                 <param name="max-wait-time" value="0"/>***
> *
>
>                                 <param name="max-wait-time-with-no-agent"
> value="120"/>****
>
>                 </queue>****
>
>                 <queue name="support at default">****
>
>                                 <param name="strategy"
> value="agent-with-least-talk-time"/>****
>
>                                 <param name="moh-sound"
> value="$${hold_music}"/>****
>
>       <!--<param name="record-template"
> value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>-->
> ****
>
>                                 <param name="time-base-score"
> value="queue"/>****
>
>                                 <param name="tier-rules-apply"
> value="false"/>****
>
>                                 <param name="tier-rule-wait-second"
> value="300"/>****
>
>                                 <param
> name="tier-rule-wait-multiply-level" value="true"/>****
>
>                                 <param name="tier-rule-no-agent-no-wait"
> value="false"/>****
>
>                                 <param name="discard-abandoned-after"
> value="60"/>****
>
>                                 <param name="abandoned-resume-allowed"
> value="false"/>****
>
>                                 <param name="max-wait-time" value="0"/>***
> *
>
>                                 <param name="max-wait-time-with-no-agent"
> value="120"/>****
>
>                 </queue>****
>
>                 <queue name="hr at default">****
>
>                                 <param name="strategy"
> value="agent-with-least-talk-time"/>****
>
>                                 <param name="moh-sound"
> value="$${hold_music}"/>****
>
>       <!--<param name="record-template"
> value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>-->
> ****
>
>                                 <param name="time-base-score"
> value="queue"/>****
>
>                                 <param name="tier-rules-apply"
> value="false"/>****
>
>                                 <param name="tier-rule-wait-second"
> value="300"/>****
>
>                                 <param
> name="tier-rule-wait-multiply-level" value="true"/>****
>
>                                 <param name="tier-rule-no-agent-no-wait"
> value="false"/>****
>
>                                 <param name="discard-abandoned-after"
> value="60"/>****
>
>                                 <param name="abandoned-resume-allowed"
> value="false"/>****
>
>                                 <param name="max-wait-time" value="0"/>***
> *
>
>                                 <param name="max-wait-time-with-no-agent"
> value="120"/>****
>
>                 </queue>****
>
>   </queues>****
>
>  ****
>
> <!-- WARNING: Configuration of XML Agents will be updated into the DB upon
> restart. -->****
>
> <!-- WARNING: Configuration of XML Tiers will reset the level and position
> if those were supplied. -->****
>
> <!-- WARNING: Agents and Tiers XML config shouldn't be used in a multi FS
> shared DB setup (Not currently supported anyway) -->****
>
>   <agents>****
>
>     <agent name="1001" type="callback"
> contact="[call_timeout=10]user/1001" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>      <agent name="1002" type="callback"
> contact="[call_timeout=10]user/1002" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>      <agent name="1003" type="callback"
> contact="[call_timeout=10]user/1003" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>      <agent name="1004" type="callback"
> contact="[call_timeout=10]user/1004" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>      <agent name="1005" type="callback"
> contact="[call_timeout=10]user/1005" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>   </agents>****
>
>   <tiers>****
>
>     <!-- If no level or position is provided, they will default to 1.  You
> should do this to keep db value on restart. -->****
>
>                 <!-- agent 1001 is in all three queues -->****
>
>                 <tier agent="1001" queue="sales at default" level="1"
> position="1"/>****
>
>                 <tier agent="1001" queue="support at default" level="1"
> position="1"/>****
>
>                 <tier agent="1001" queue="hr at default" level="1"
> position="1"/>****
>
>  ****
>
>                 <tier agent="1002" queue="sales at default" level="1"
> position="1"/>****
>
>  ****
>
>                 <tier agent="1003" queue="support at default" level="1"
> position="1"/>****
>
>  ****
>
>                 <tier agent="1004" queue="sales at default" level="1"
> position="1"/>****
>
>                 <tier agent="1004" queue="support at default" level="1"
> position="1"/>****
>
>  ****
>
>                 <tier agent="1005" queue="hr at default" level="1"
> position="1"/>****
>
>   </tiers>****
>
>  ****
>
> </configuration>****
>
>  ****
>
> When I call from extension 1000 and press 2 for sales, call is routed back
> to extension 1000, which is not even in agents? list. Why is it happening?
> ****
>
>  ****
>
> Regards****
>
>  ****
>
> Nitin****
>
> ** **
>
>
> Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL.
>
>
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From vbvbrj at gmail.com  Fri Aug 31 18:40:49 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Fri, 31 Aug 2012 17:40:49 +0300
Subject: [Freeswitch-users] Recording problem
In-Reply-To: <CALvF6vCS8tW_4_4ecGhrcU6ia+8A0BoAHSfquCLK2VnFpyHx4Q@mail.gmail.com>
References: <CALvF6vCS8tW_4_4ecGhrcU6ia+8A0BoAHSfquCLK2VnFpyHx4Q@mail.gmail.com>
Message-ID: <5040CCF1.9000200@gmail.com>

On 31.08.2012 17:07, Jayesh Nambiar wrote:
> Hi,
> I had this same problem and I don't think it is related to
> mod_callcenter. I had also sent a post in regards to this here:
> http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/085552.html

It's a common issue. I had too, so I added to global pref codec list the 
codecs that will be used by phones. Also check phone signal standard. 
This must be posted to gira.

-- 
Mimiko desu.


From msc at freeswitch.org  Fri Aug 31 18:59:23 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 31 Aug 2012 07:59:23 -0700
Subject: [Freeswitch-users] endpoint /user and sofia_contact change?
In-Reply-To: <CALNZuwmsOvxTU3aEy=07_GNzU7_UtD+KNkPKwc70nz1q9YyqAg@mail.gmail.com>
References: <CALNZuwnv86enuhdOm57zSwF7G-D9U4gFnUBZR8R+QWjC5Wu4zw@mail.gmail.com>
	<CAC-LwPMoGO2GCeMxC3Yw6-FV6wzwA6LvTyFKbmWt9O8s7Ua9pw@mail.gmail.com>
	<CALNZuwmmm_3yo2tdYT_58M7Qgbz7QiRzmf42kpCwta5psD4YFg@mail.gmail.com>
	<CAKbxfG9QYWebJ0RkH21yYxD2sLjjhJ+-7PYNVLJ8aNe2zU6kFA@mail.gmail.com>
	<CALNZuwmsOvxTU3aEy=07_GNzU7_UtD+KNkPKwc70nz1q9YyqAg@mail.gmail.com>
Message-ID: <CAKzWOxVaPhfiWyVVVPrfDXWaLUS9iwhKm+ZR+hT1qO8TWtRhvg@mail.gmail.com>

Thanks for updating the wiki on this.
-MC

On Fri, Aug 31, 2012 at 12:44 AM, Avi Marcus <avi at avimarcus.net> wrote:

> Ah, ok. This needs to be noted somewhere....
>
> Also, how does this affect bridging to "user", e.g. user/1000 ? I've never
> seen the profile in one of those.
>
> -Avi
>
>
> On Fri, Aug 31, 2012 at 3:29 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Reverse that, a bug was fixed that may now appear as a bug, add the
>> special profile name * in your sofia_contact call so it searches all
>> profiles...
>>
>> sofia_contact */user at domain.com
>> On Aug 29, 2012 10:36 AM, "Avi Marcus" <avi at avimarcus.net> wrote:
>>
>>> freeswitch at default> sofia_contact 1000
>>> error/user_not_registered
>>>
>>> freeswitch at default> sofia_contact 1000 at sip.mydomain.com
>>> error/user_not_registered
>>>
>>> freeswitch at default> sofia_contact internal/1000 at sip.mydomain.com
>>> sofia/internal/sip:1000 at 192.168.255.103:5077,sofia/internal/
>>> sip:1000 at 192.168.255.103:5078
>>> ,sofia/internal/sip:1000 at 192.168.255.103:32998;ob
>>>
>>> The first 2 worked before. I suppose this is a bug.
>>>
>>> -Avi
>>>
>>>
>>> On Wed, Aug 29, 2012 at 6:23 PM, Vik Killa <vipkilla at gmail.com> wrote:
>>>
>>>> does 'sofia status profile <profile_name> reg' show them as registered?
>>>> try registering the handsets or flushing sofia registrations and
>>>> re-registering phone.
>>>>
>>>> On Wed, Aug 29, 2012 at 11:07 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>>> > I think I had 1.2 release yesterday and I would bridge to user/1000
>>>> or user
>>>> > 1000 at mydomain.com and/or do sofia_contact(1000) and all is fine.
>>>> > However, since building on master in GIT last night: FreeSWITCH
>>>> Version
>>>> > 1.3.0+git~20120828T230559Z~f1d201f03b (1.3.0; git at commit
>>>> f1d201f03b on
>>>> > Tue, 28 Aug 2012 23:05:59 Z)
>>>> >
>>>> > They return not registered. I have to do sofia_contact internal/1000.
>>>> I'm
>>>> > not sure how to use user/ I replaced it with
>>>> > ${sofia_contact(internal/1000)}.
>>>> > Was this an intentional change?
>>>> > Is there some list of breaking changes somewhere?
>>>> >
>>>> > -Avi
>>>> >
>>>> >
>>>> _________________________________________________________________________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > 
>>>> > 
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://wiki.freeswitch.org
>>>> > http://www.cluecon.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
>>>> >
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From ntomer at newgen.co.in  Fri Aug 31 19:07:24 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Fri, 31 Aug 2012 20:37:24 +0530
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded
	to	caller itself
In-Reply-To: <CALvF6vBeLSrW70vsPeAK0V4h-28Cr1y+_fD0wmx-0TLr1hTVrQ@mail.gmail.com>
References: <CALvF6vBeLSrW70vsPeAK0V4h-28Cr1y+_fD0wmx-0TLr1hTVrQ@mail.gmail.com>
Message-ID: <01bb01cd878a$53f6c410$fbe44c30$@co.in>

You are right Jayesh, agent 1000 s being listed from CLI. But it is not in
callcenter.cong.xml. Neither I have added it from CLI. Where it is coming
from?

 

Thanks

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jayesh
Nambiar
Sent: Friday, August 31, 2012 8:07 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_callcenter: Call being forwarded to
caller itself

 

Hi Nitin,

In that case all you have to confirm is that you have not set the contact of
any agent as 1000. Keep in mind that the agents can be pushed into
freeswitch memory from API commands also. Basically you can use FS_CLI to
add agents and set their contacts even without mentioning those agents in
the XML file. So just be sure that you do not have any agent's contact set
to 1000 and also check from FS_CLI issuing this command: "callcenter_config
agent list" to make sure that none of the agent has contact set to 1000.

 

--- Jayesh

 

 

Hi Jayesh,

 

Call is being redirected to call center because it is working other two
options. Even for "sales" the call is redirected to caller(extn 1000) but
when it is declined from there, it works as desired.

 

Thanks

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jayesh
Nambiar
Sent: Friday, August 31, 2012 6:55 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_callcenter: Call being forwarded to
caller itself

 

Nitin, 

If this is   your exact configuration, it is simply not possible that the
call gets routed back to same extension unless there is something configured
in the dialplan.

Look at the console in loglevel 7 mode and check if the call goes into
callcenter module or no? If no, check the flow of your dialplan and figure
out what is taking the call back to the calling extension.

 

--- Jayesh 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin
Tomer

Sent: Wednesday, August 29, 2012 3:39 PM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded to caller
itself

 

Hi,

 

I've configure mod_callcenter on Ubuntu 12.0.4. I've 5 extensions working on
6 soft phones - 1000 to 1005. 1000 is used to call the IVR number, rest 5
are agents.

 

The dial plan looks like -

 

<extension name="ivr_nitin">

      <condition field="destination_number" expression="^5002$">

        <action application="answer"/>

        <action application="sleep" data="500"/>

                <action application="ivr" data="nitin_ivr"/>

      </condition>

    </extension>

<!-- For Call Center -->

<!-- Newgen Support -->

       <extension>

          <condition field="destination_number" expression="^(450)$">

              <action application="set" data="caller_id_name=Newgen Support"
/>

              <action application="set" data="call_timeout=60" />

              <action application="set" data="originate_timeout=60" />

              <action application="callcenter" data="support at default"/>

           </condition>

        </extension>

 

<!-- Newgen Sales -->

        <extension>

           <condition field="destination_number" expression="^(451)$">

               <action application="set" data="caller_id_name=Newgen Sales"
/>

               <action application="set" data="call_timeout=60" />

               <action application="set" data="originate_timeout=60" />

               <action application="callcenter" data="sales at default"/>

           </condition>

        </extension>

 

<!-- Newgen Human Resources -->

        <extension>

           <condition field="destination_number" expression="^(452)$">

               <action application="set" data="caller_id_name=Newgen Human
Resources" />

                       <action application="set" data="call_timeout=60" />

               <action application="set" data="originate_timeout=60" />

               <action application="callcenter" data="hr at default"/>

           </condition>

        </extension>

 

IVR.xml -

 

<include>

        <menu name="nitin_ivr"

                greet-long="say:Welcome to Newgen Software Technologies
Limited. Press 1 for support, 2 for sales or 3 for human resource"

                greet-short="say:Welcome to Newgen. Press 1 for support, 2
for sales or 3 for human resources"

                invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"

                exit-sound="voicemail/vm-goodbye.wav"

                confirm-macro=""

                confirm-key=""

                tts-engine="flite"

                tts-voice="slt"

                confirm-attempts="3"

                timeout="3000"

                inter-digit-timeout="2000"

                max-failures="3"

                max-timeouts="3"

                digit-len="4">

 

                <entry action="menu-exec-app" digits="1" param="transfer 450
XML default"/>    <!-- Support -->

                <entry action="menu-exec-app" digits="2" param="transfer 451
XML default"/>    <!-- Sales -->

                <entry action="menu-exec-app" digits="3" param="transfer 452
XML default"/>    <!-- Human Resources -->

 

        </menu>

</include>

 

And callcenter.conf.xml -

 

<configuration name="callcenter.conf" description="CallCenter">

  <settings>

    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->

    <!--<param name="dbname" value="/dev/shm/callcenter.db"/>-->

  </settings>

 

    <queues>

                <queue name="sales at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/sales/${strftime(%Y-%m-%d-%H-%M-%S)}.${destin
ation_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="14400"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

                <queue name="support at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${dest
ination_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="60"/>

                                <param name="abandoned-resume-allowed"
value="false"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

                <queue name="hr at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${dest
ination_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="60"/>

                                <param name="abandoned-resume-allowed"
value="false"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

  </queues>

 

<!-- WARNING: Configuration of XML Agents will be updated into the DB upon
restart. -->

<!-- WARNING: Configuration of XML Tiers will reset the level and position
if those were supplied. -->

<!-- WARNING: Agents and Tiers XML config shouldn't be used in a multi FS
shared DB setup (Not currently supported anyway) -->

  <agents>

    <agent name="1001" type="callback" contact="[call_timeout=10]user/1001"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1002" type="callback" contact="[call_timeout=10]user/1002"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1003" type="callback" contact="[call_timeout=10]user/1003"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1004" type="callback" contact="[call_timeout=10]user/1004"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1005" type="callback" contact="[call_timeout=10]user/1005"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

  </agents>

  <tiers>

    <!-- If no level or position is provided, they will default to 1.  You
should do this to keep db value on restart. -->

                <!-- agent 1001 is in all three queues -->

                <tier agent="1001" queue="sales at default" level="1"
position="1"/>

                <tier agent="1001" queue="support at default" level="1"
position="1"/>

                <tier agent="1001" queue="hr at default" level="1"
position="1"/>

 

                <tier agent="1002" queue="sales at default" level="1"
position="1"/>

 

                <tier agent="1003" queue="support at default" level="1"
position="1"/>

 

                <tier agent="1004" queue="sales at default" level="1"
position="1"/>

                <tier agent="1004" queue="support at default" level="1"
position="1"/>

 

                <tier agent="1005" queue="hr at default" level="1"
position="1"/>

  </tiers>

 

</configuration>

 

When I call from extension 1000 and press 2 for sales, call is routed back
to extension 1000, which is not even in agents' list. Why is it happening?

 

Regards

 

Nitin

 

 


Disclaimer :- This e-mail and any attachment may contain confidential,
proprietary or legally privileged information. If you are not the original
intended recipient and have erroneously received this message, you are
prohibited from using, copying, altering or disclosing the content of this
message. Please delete it immediately and notify the sender. Newgen Software
Technologies Ltd (NSTL) accepts no responsibilities for loss or damage
arising from the use of the information transmitted by this email including
damages from virus and further acknowledges that no binding nature of the
message shall be implied or assumed unless the sender does so expressly with
due authority of NSTL. 


 

 


Disclaimer :- This e-mail and any attachment may contain confidential, proprietary or legally privileged information. If you are not the original intended recipient and have erroneously received this message, you are prohibited from using, copying, altering or disclosing the content of this message. Please delete it immediately and notify the sender. Newgen Software Technologies Ltd (NSTL)  accepts no responsibilities for loss or damage arising from the use of the information transmitted by this email including damages from virus and further acknowledges that no binding nature of the message shall be implied or assumed unless the sender does so expressly with due authority of NSTL.
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From msc at freeswitch.org  Fri Aug 31 19:05:42 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 31 Aug 2012 08:05:42 -0700
Subject: [Freeswitch-users] Asking a Question (a.k.a. Gabe must be
	getting old)
In-Reply-To: <CAMwZ-tJ7Asfd_MSWqNjCdEsSF=DOx9daGZ_T5CbWzoxfTb2UfA@mail.gmail.com>
References: <CAMwZ-tJ7Asfd_MSWqNjCdEsSF=DOx9daGZ_T5CbWzoxfTb2UfA@mail.gmail.com>
Message-ID: <CAKzWOxVKP81wj=86g4_gvRp=d2SkTwA8X_QF5hY2eruTz+iLvg@mail.gmail.com>

On Thu, Aug 30, 2012 at 11:08 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> This should be required reading before posting questions to the list:
>
> http://www.catb.org/esr/faqs/smart-questions.html#intro
>
> Special thanks to those people on the list who take special care to
> ask *smart* questions and those who provide valuable answers. You know
> who you are ;)
>
> +1 - this is actually very handy
-MC
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From vipkilla at gmail.com  Fri Aug 31 19:14:54 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 31 Aug 2012 11:14:54 -0400
Subject: [Freeswitch-users] endpoint /user and sofia_contact change?
In-Reply-To: <CAKzWOxVaPhfiWyVVVPrfDXWaLUS9iwhKm+ZR+hT1qO8TWtRhvg@mail.gmail.com>
References: <CALNZuwnv86enuhdOm57zSwF7G-D9U4gFnUBZR8R+QWjC5Wu4zw@mail.gmail.com>
	<CAC-LwPMoGO2GCeMxC3Yw6-FV6wzwA6LvTyFKbmWt9O8s7Ua9pw@mail.gmail.com>
	<CALNZuwmmm_3yo2tdYT_58M7Qgbz7QiRzmf42kpCwta5psD4YFg@mail.gmail.com>
	<CAKbxfG9QYWebJ0RkH21yYxD2sLjjhJ+-7PYNVLJ8aNe2zU6kFA@mail.gmail.com>
	<CALNZuwmsOvxTU3aEy=07_GNzU7_UtD+KNkPKwc70nz1q9YyqAg@mail.gmail.com>
	<CAKzWOxVaPhfiWyVVVPrfDXWaLUS9iwhKm+ZR+hT1qO8TWtRhvg@mail.gmail.com>
Message-ID: <CAC-LwPP-6ZBoN8q7+JfSrfNLwq8rVm2VsB774SdL=aHC2b7qBg@mail.gmail.com>

Which wiki page was updated?

On Fri, Aug 31, 2012 at 10:59 AM, Michael Collins <msc at freeswitch.org> wrote:
> Thanks for updating the wiki on this.
> -MC
>


From krice at freeswitch.org  Fri Aug 31 19:21:52 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 31 Aug 2012 10:21:52 -0500
Subject: [Freeswitch-users] Asking a Question (a.k.a. Gabe must be
	getting old)
In-Reply-To: <CAKzWOxVKP81wj=86g4_gvRp=d2SkTwA8X_QF5hY2eruTz+iLvg@mail.gmail.com>
References: <CAMwZ-tJ7Asfd_MSWqNjCdEsSF=DOx9daGZ_T5CbWzoxfTb2UfA@mail.gmail.com>
	<CAKzWOxVKP81wj=86g4_gvRp=d2SkTwA8X_QF5hY2eruTz+iLvg@mail.gmail.com>
Message-ID: <CAPcNkdfPenKFWF_yauwj6-XpiqJ_PjpOMCA_qN09cdHkZZyJdg@mail.gmail.com>

to bad we cant get more people to read this before they post...

also people, really... when googling

site:lists.freeswitch.org whatever you want to know
site:wiki.freeswitch.org find it on the wiki




On Fri, Aug 31, 2012 at 10:05 AM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Thu, Aug 30, 2012 at 11:08 PM, Gabriel Gunderson <gabe at gundy.org>wrote:
>
>> This should be required reading before posting questions to the list:
>>
>> http://www.catb.org/esr/faqs/smart-questions.html#intro
>>
>> Special thanks to those people on the list who take special care to
>> ask *smart* questions and those who provide valuable answers. You know
>> who you are ;)
>>
>> +1 - this is actually very handy
> -MC
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 avi at avimarcus.net  Fri Aug 31 19:24:40 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 31 Aug 2012 18:24:40 +0300
Subject: [Freeswitch-users] endpoint /user and sofia_contact change?
In-Reply-To: <CAC-LwPP-6ZBoN8q7+JfSrfNLwq8rVm2VsB774SdL=aHC2b7qBg@mail.gmail.com>
References: <CALNZuwnv86enuhdOm57zSwF7G-D9U4gFnUBZR8R+QWjC5Wu4zw@mail.gmail.com>
	<CAC-LwPMoGO2GCeMxC3Yw6-FV6wzwA6LvTyFKbmWt9O8s7Ua9pw@mail.gmail.com>
	<CALNZuwmmm_3yo2tdYT_58M7Qgbz7QiRzmf42kpCwta5psD4YFg@mail.gmail.com>
	<CAKbxfG9QYWebJ0RkH21yYxD2sLjjhJ+-7PYNVLJ8aNe2zU6kFA@mail.gmail.com>
	<CALNZuwmsOvxTU3aEy=07_GNzU7_UtD+KNkPKwc70nz1q9YyqAg@mail.gmail.com>
	<CAKzWOxVaPhfiWyVVVPrfDXWaLUS9iwhKm+ZR+hT1qO8TWtRhvg@mail.gmail.com>
	<CAC-LwPP-6ZBoN8q7+JfSrfNLwq8rVm2VsB774SdL=aHC2b7qBg@mail.gmail.com>
Message-ID: <CALNZuwnyUrHEY0YwdPzhU12+tB6iM7pSG4HLX5-uXu4cpCgGrw@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Function_sofia_contact was updated.

I'm not sure how this affects bridging to user/1000. I've replaced that
with ${sofia_contact(*/1000)} for now.

-Avi


On Fri, Aug 31, 2012 at 6:14 PM, Vik Killa <vipkilla at gmail.com> wrote:

> Which wiki page was updated?
>
> On Fri, Aug 31, 2012 at 10:59 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Thanks for updating the wiki on this.
> > -MC
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayesh.voip at gmail.com  Fri Aug 31 19:25:43 2012
From: jayesh.voip at gmail.com (Jayesh Nambiar)
Date: Fri, 31 Aug 2012 20:55:43 +0530
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded to
	caller itself
Message-ID: <CALvF6vAoMBhg0N_jhiEeUFRxHdB2Saiv5iPf=uFNBmWDvxR5XQ@mail.gmail.com>

>
> Nitin,

It is quite possible that you must have added it sometime earlier and only
deleted it from the XML and not from the freeswitch memory. Freeswitch
callcenter module basically maintains an SQLite DB where it stores all
these values. You will have to remove it from the freeswitch memory.
Execute this from freeswitch CLI:
callcenter_config agent del <agent_name>

--- Jayesh


> You are right Jayesh, agent 1000 s being listed from CLI. But it is not in
> callcenter.cong.xml. Neither I have added it from CLI. Where it is coming
> from?
>
> ** **
>
> Thanks****
>
> ** **
>
> Nitin****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayesh
> Nambiar
> *Sent:* Friday, August 31, 2012 8:07 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] mod_callcenter: Call being forwarded to
> caller itself****
>
> ** **
>
> Hi Nitin,****
>
> In that case all you have to confirm is that you have not set the contact
> of any agent as 1000. Keep in mind that the agents can be pushed into
> freeswitch memory from API commands also. Basically you can use FS_CLI to
> add agents and set their contacts even without mentioning those agents in
> the XML file. So just be sure that you do not have any agent's contact set
> to 1000 and also check from FS_CLI issuing this command: "callcenter_config
> agent list" to make sure that none of the agent has contact set to 1000.**
> **
>
> ** **
>
> --- Jayesh****
>
> ** **
>
>  ****
>
>  Hi Jayesh,****
>
>  ****
>
> Call is being redirected to call center because it is working other two
> options. Even for ?sales? the call is redirected to caller(extn 1000) but
> when it is declined from there, it works as desired.****
>
>  ****
>
> Thanks****
>
>  ****
>
> Nitin****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayesh
> Nambiar
> *Sent:* Friday, August 31, 2012 6:55 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] mod_callcenter: Call being forwarded to
> caller itself****
>
>  ****
>
> Nitin, ****
>
> If this is   your exact configuration, it is simply not possible that the
> call gets routed back to same extension unless there is something
> configured in the dialplan.****
>
>  Look at the console in loglevel 7 mode and check if the call goes into
> callcenter module or no? If no, check the flow of your dialplan and figure
> out what is taking the call back to the calling extension.****
>
>  ****
>
> --- Jayesh ****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nitin Tomer*
> ***
>
> *Sent:* Wednesday, August 29, 2012 3:39 PM
> *To:* 'FreeSWITCH Users Help'
> *Subject:* [Freeswitch-users] mod_callcenter: Call being forwarded to
> caller itself****
>
>  ****
>
> Hi,****
>
>  ****
>
> I?ve configure mod_callcenter on Ubuntu 12.0.4. I?ve 5 extensions working
> on 6 soft phones ? 1000 to 1005. 1000 is used to call the IVR number, rest
> 5 are agents.****
>
>  ****
>
> The dial plan looks like ?****
>
>  ****
>
> <extension name="ivr_nitin">****
>
>       <condition field="destination_number" expression="^5002$">****
>
>         <action application="answer"/>****
>
>         <action application="sleep" data="500"/>****
>
>                 <action application="ivr" data="nitin_ivr"/>****
>
>       </condition>****
>
>     </extension>****
>
> <!-- For Call Center -->****
>
> <!-- Newgen Support -->****
>
>        <extension>****
>
>           <condition field="destination_number" expression="^(450)$">****
>
>               <action application="set" data="caller_id_name=Newgen
> Support" />****
>
>               <action application="set" data="call_timeout=60" />****
>
>               <action application="set" data="originate_timeout=60" />****
>
>               <action application="callcenter" data="support at default"/>***
> *
>
>            </condition>****
>
>         </extension>****
>
>  ****
>
> <!-- Newgen Sales -->****
>
>         <extension>****
>
>            <condition field="destination_number" expression="^(451)$">****
>
>                <action application="set" data="caller_id_name=Newgen
> Sales" />****
>
>                <action application="set" data="call_timeout=60" />****
>
>                <action application="set" data="originate_timeout=60" />***
> *
>
>                <action application="callcenter" data="sales at default"/>****
>
>            </condition>****
>
>         </extension>****
>
>  ****
>
> <!-- Newgen Human Resources -->****
>
>         <extension>****
>
>            <condition field="destination_number" expression="^(452)$">****
>
>                <action application="set" data="caller_id_name=Newgen Human
> Resources" />****
>
>                        <action application="set" data="call_timeout=60" />
> ****
>
>                <action application="set" data="originate_timeout=60" />***
> *
>
>                <action application="callcenter" data="hr at default"/>****
>
>            </condition>****
>
>         </extension>****
>
>  ****
>
> IVR.xml ?****
>
>  ****
>
> <include>****
>
>         <menu name="nitin_ivr"****
>
>                 greet-long="say:Welcome to Newgen Software Technologies
> Limited. Press 1 for support, 2 for sales or 3 for human resource"****
>
>                 greet-short="say:Welcome to Newgen. Press 1 for support, 2
> for sales or 3 for human resources"****
>
>                 invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"****
>
>                 exit-sound="voicemail/vm-goodbye.wav"****
>
>                 confirm-macro=""****
>
>                 confirm-key=""****
>
>                 tts-engine="flite"****
>
>                 tts-voice="slt"****
>
>                 confirm-attempts="3"****
>
>                 timeout="3000"****
>
>                 inter-digit-timeout="2000"****
>
>                 max-failures="3"****
>
>                 max-timeouts="3"****
>
>                 digit-len="4">****
>
>  ****
>
>                 <entry action="menu-exec-app" digits="1" param="transfer
> 450 XML default"/>    <!-- Support -->****
>
>                 <entry action="menu-exec-app" digits="2" param="transfer
> 451 XML default"/>    <!-- Sales -->****
>
>                 <entry action="menu-exec-app" digits="3" param="transfer
> 452 XML default"/>    <!-- Human Resources -->****
>
>  ****
>
>         </menu>****
>
> </include>****
>
>  ****
>
> And callcenter.conf.xml ?****
>
>  ****
>
> <configuration name="callcenter.conf" description="CallCenter">****
>
>   <settings>****
>
>     <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->****
>
>     <!--<param name="dbname" value="/dev/shm/callcenter.db"/>-->****
>
>   </settings>****
>
>  ****
>
>     <queues>****
>
>                 <queue name="sales at default">****
>
>                                 <param name="strategy"
> value="agent-with-least-talk-time"/>****
>
>                                 <param name="moh-sound"
> value="$${hold_music}"/>****
>
>       <!--<param name="record-template"
> value="$${base_dir}/recordings/sales/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>-->
> ****
>
>                                 <param name="time-base-score"
> value="queue"/>****
>
>                                 <param name="tier-rules-apply"
> value="false"/>****
>
>                                 <param name="tier-rule-wait-second"
> value="300"/>****
>
>                                 <param
> name="tier-rule-wait-multiply-level" value="true"/>****
>
>                                 <param name="tier-rule-no-agent-no-wait"
> value="false"/>****
>
>                                 <param name="discard-abandoned-after"
> value="14400"/>****
>
>                                 <param name="max-wait-time" value="0"/>***
> *
>
>                                 <param name="max-wait-time-with-no-agent"
> value="120"/>****
>
>                 </queue>****
>
>                 <queue name="support at default">****
>
>                                 <param name="strategy"
> value="agent-with-least-talk-time"/>****
>
>                                 <param name="moh-sound"
> value="$${hold_music}"/>****
>
>       <!--<param name="record-template"
> value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>-->
> ****
>
>                                 <param name="time-base-score"
> value="queue"/>****
>
>                                 <param name="tier-rules-apply"
> value="false"/>****
>
>                                 <param name="tier-rule-wait-second"
> value="300"/>****
>
>                                 <param
> name="tier-rule-wait-multiply-level" value="true"/>****
>
>                                 <param name="tier-rule-no-agent-no-wait"
> value="false"/>****
>
>                                 <param name="discard-abandoned-after"
> value="60"/>****
>
>                                 <param name="abandoned-resume-allowed"
> value="false"/>****
>
>                                 <param name="max-wait-time" value="0"/>***
> *
>
>                                 <param name="max-wait-time-with-no-agent"
> value="120"/>****
>
>                 </queue>****
>
>                 <queue name="hr at default">****
>
>                                 <param name="strategy"
> value="agent-with-least-talk-time"/>****
>
>                                 <param name="moh-sound"
> value="$${hold_music}"/>****
>
>       <!--<param name="record-template"
> value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>-->
> ****
>
>                                 <param name="time-base-score"
> value="queue"/>****
>
>                                 <param name="tier-rules-apply"
> value="false"/>****
>
>                                 <param name="tier-rule-wait-second"
> value="300"/>****
>
>                                 <param
> name="tier-rule-wait-multiply-level" value="true"/>****
>
>                                 <param name="tier-rule-no-agent-no-wait"
> value="false"/>****
>
>                                 <param name="discard-abandoned-after"
> value="60"/>****
>
>                                 <param name="abandoned-resume-allowed"
> value="false"/>****
>
>                                 <param name="max-wait-time" value="0"/>***
> *
>
>                                 <param name="max-wait-time-with-no-agent"
> value="120"/>****
>
>                 </queue>****
>
>   </queues>****
>
>  ****
>
> <!-- WARNING: Configuration of XML Agents will be updated into the DB upon
> restart. -->****
>
> <!-- WARNING: Configuration of XML Tiers will reset the level and position
> if those were supplied. -->****
>
> <!-- WARNING: Agents and Tiers XML config shouldn't be used in a multi FS
> shared DB setup (Not currently supported anyway) -->****
>
>   <agents>****
>
>     <agent name="1001" type="callback"
> contact="[call_timeout=10]user/1001" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>      <agent name="1002" type="callback"
> contact="[call_timeout=10]user/1002" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>      <agent name="1003" type="callback"
> contact="[call_timeout=10]user/1003" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>      <agent name="1004" type="callback"
> contact="[call_timeout=10]user/1004" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>      <agent name="1005" type="callback"
> contact="[call_timeout=10]user/1005" status="Available" max-no-answer="3"
> wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />****
>
>   </agents>****
>
>   <tiers>****
>
>     <!-- If no level or position is provided, they will default to 1.  You
> should do this to keep db value on restart. -->****
>
>                 <!-- agent 1001 is in all three queues -->****
>
>                 <tier agent="1001" queue="sales at default" level="1"
> position="1"/>****
>
>                 <tier agent="1001" queue="support at default" level="1"
> position="1"/>****
>
>                 <tier agent="1001" queue="hr at default" level="1"
> position="1"/>****
>
>  ****
>
>                 <tier agent="1002" queue="sales at default" level="1"
> position="1"/>****
>
>  ****
>
>                 <tier agent="1003" queue="support at default" level="1"
> position="1"/>****
>
>  ****
>
>                 <tier agent="1004" queue="sales at default" level="1"
> position="1"/>****
>
>                 <tier agent="1004" queue="support at default" level="1"
> position="1"/>****
>
>  ****
>
>                 <tier agent="1005" queue="hr at default" level="1"
> position="1"/>****
>
>   </tiers>****
>
>  ****
>
> </configuration>****
>
>  ****
>
> When I call from extension 1000 and press 2 for sales, call is routed back
> to extension 1000, which is not even in agents? list. Why is it happening?
> ****
>
>  ****
>
> Regards****
>
>  ****
>
> Nitin****
>
>  ****
>
>  ** **
>  Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
> message. Please delete it immediately and notify the sender. Newgen
> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
> damage arising from the use of the information transmitted by this email
> including damages from virus and further acknowledges that no binding
> nature of the message shall be implied or assumed unless the sender does so
> expressly with due authority of NSTL. ****
>
> ** **
>
>  ** **
>
> Disclaimer :- This e-mail and any attachment may contain confidential,
> proprietary or legally privileged information. If you are not the original
> intended recipient and have erroneously received this message, you are
> prohibited from using, copying, altering or disclosing the content of this
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> Software Technologies Ltd (NSTL) accepts no responsibilities for loss or
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From msc at freeswitch.org  Fri Aug 31 19:30:28 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 31 Aug 2012 08:30:28 -0700
Subject: [Freeswitch-users] endpoint /user and sofia_contact change?
In-Reply-To: <CAC-LwPP-6ZBoN8q7+JfSrfNLwq8rVm2VsB774SdL=aHC2b7qBg@mail.gmail.com>
References: <CALNZuwnv86enuhdOm57zSwF7G-D9U4gFnUBZR8R+QWjC5Wu4zw@mail.gmail.com>
	<CAC-LwPMoGO2GCeMxC3Yw6-FV6wzwA6LvTyFKbmWt9O8s7Ua9pw@mail.gmail.com>
	<CALNZuwmmm_3yo2tdYT_58M7Qgbz7QiRzmf42kpCwta5psD4YFg@mail.gmail.com>
	<CAKbxfG9QYWebJ0RkH21yYxD2sLjjhJ+-7PYNVLJ8aNe2zU6kFA@mail.gmail.com>
	<CALNZuwmsOvxTU3aEy=07_GNzU7_UtD+KNkPKwc70nz1q9YyqAg@mail.gmail.com>
	<CAKzWOxVaPhfiWyVVVPrfDXWaLUS9iwhKm+ZR+hT1qO8TWtRhvg@mail.gmail.com>
	<CAC-LwPP-6ZBoN8q7+JfSrfNLwq8rVm2VsB774SdL=aHC2b7qBg@mail.gmail.com>
Message-ID: <CAKzWOxWiinze-O3tjMEtxtazxJM_mP6Vk2vYGysBPwppjwq6pw@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Function_sofia_contact

On Fri, Aug 31, 2012 at 8:14 AM, Vik Killa <vipkilla at gmail.com> wrote:

> Which wiki page was updated?
>
> On Fri, Aug 31, 2012 at 10:59 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Thanks for updating the wiki on this.
> > -MC
> >
>
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From vbvbrj at gmail.com  Fri Aug 31 19:31:12 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Fri, 31 Aug 2012 18:31:12 +0300
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded to
 caller itself
In-Reply-To: <CALvF6vAoMBhg0N_jhiEeUFRxHdB2Saiv5iPf=uFNBmWDvxR5XQ@mail.gmail.com>
References: <CALvF6vAoMBhg0N_jhiEeUFRxHdB2Saiv5iPf=uFNBmWDvxR5XQ@mail.gmail.com>
Message-ID: <5040D8C0.7000401@gmail.com>

On 31.08.2012 18:25, Jayesh Nambiar wrote:
>     Nitin,
>
> It is quite possible that you must have added it sometime earlier and
> only deleted it from the XML and not from the freeswitch memory.
> Freeswitch callcenter module basically maintains an SQLite DB where it
> stores all these values. You will have to remove it from the freeswitch
> memory. Execute this from freeswitch CLI:
> callcenter_config agent del <agent_name>

On fs_cli use:
callcenter_config tier list
to see the tiers which is acting.

Also post the log to see the dialplan passing.

-- 
Mimiko desu.


From anthony.minessale at gmail.com  Fri Aug 31 19:32:31 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 31 Aug 2012 10:32:31 -0500
Subject: [Freeswitch-users] endpoint /user and sofia_contact change?
In-Reply-To: <CALNZuwnyUrHEY0YwdPzhU12+tB6iM7pSG4HLX5-uXu4cpCgGrw@mail.gmail.com>
References: <CALNZuwnv86enuhdOm57zSwF7G-D9U4gFnUBZR8R+QWjC5Wu4zw@mail.gmail.com>
	<CAC-LwPMoGO2GCeMxC3Yw6-FV6wzwA6LvTyFKbmWt9O8s7Ua9pw@mail.gmail.com>
	<CALNZuwmmm_3yo2tdYT_58M7Qgbz7QiRzmf42kpCwta5psD4YFg@mail.gmail.com>
	<CAKbxfG9QYWebJ0RkH21yYxD2sLjjhJ+-7PYNVLJ8aNe2zU6kFA@mail.gmail.com>
	<CALNZuwmsOvxTU3aEy=07_GNzU7_UtD+KNkPKwc70nz1q9YyqAg@mail.gmail.com>
	<CAKzWOxVaPhfiWyVVVPrfDXWaLUS9iwhKm+ZR+hT1qO8TWtRhvg@mail.gmail.com>
	<CAC-LwPP-6ZBoN8q7+JfSrfNLwq8rVm2VsB774SdL=aHC2b7qBg@mail.gmail.com>
	<CALNZuwnyUrHEY0YwdPzhU12+tB6iM7pSG4HLX5-uXu4cpCgGrw@mail.gmail.com>
Message-ID: <CAKbxfG-s+oK-3ks1Fd+9rWDXNkwbNgW5n=eoxqD_5LqBAm-2QA@mail.gmail.com>

user endpoint uses sofia_contact internally in the config under
dial-string param


On Fri, Aug 31, 2012 at 10:24 AM, Avi Marcus <avi at avimarcus.net> wrote:
> http://wiki.freeswitch.org/wiki/Function_sofia_contact was updated.
>
> I'm not sure how this affects bridging to user/1000. I've replaced that with
> ${sofia_contact(*/1000)} for now.
>
> -Avi
>
>
> On Fri, Aug 31, 2012 at 6:14 PM, Vik Killa <vipkilla at gmail.com> wrote:
>>
>> Which wiki page was updated?
>>
>> On Fri, Aug 31, 2012 at 10:59 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > Thanks for updating the wiki on this.
>> > -MC
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From msc at freeswitch.org  Fri Aug 31 19:33:29 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 31 Aug 2012 08:33:29 -0700
Subject: [Freeswitch-users] Asking a Question (a.k.a. Gabe must be
	getting old)
In-Reply-To: <CAPcNkdfPenKFWF_yauwj6-XpiqJ_PjpOMCA_qN09cdHkZZyJdg@mail.gmail.com>
References: <CAMwZ-tJ7Asfd_MSWqNjCdEsSF=DOx9daGZ_T5CbWzoxfTb2UfA@mail.gmail.com>
	<CAKzWOxVKP81wj=86g4_gvRp=d2SkTwA8X_QF5hY2eruTz+iLvg@mail.gmail.com>
	<CAPcNkdfPenKFWF_yauwj6-XpiqJ_PjpOMCA_qN09cdHkZZyJdg@mail.gmail.com>
Message-ID: <CAKzWOxXnNQWFwdMTujLUYX6z6yk8x0XuCVSLc=kNnQgkcLcJqQ@mail.gmail.com>

FYI, I added a quickie link to the front page for the page Gabe suggested.
I think this thread has given us our topic for next week's conference call:
documenting how to find stuff. Let's talk about how we find stuff and then
write up a nice page that we can link to when people ask
vague/broad/I'm-really-lazy questions.

-MC

On Fri, Aug 31, 2012 at 8:21 AM, Ken Rice <krice at freeswitch.org> wrote:

> to bad we cant get more people to read this before they post...
>
> also people, really... when googling
>
> site:lists.freeswitch.org whatever you want to know
> site:wiki.freeswitch.org find it on the wiki
>
>
>
>
> On Fri, Aug 31, 2012 at 10:05 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Thu, Aug 30, 2012 at 11:08 PM, Gabriel Gunderson <gabe at gundy.org>wrote:
>>
>>> This should be required reading before posting questions to the list:
>>>
>>> http://www.catb.org/esr/faqs/smart-questions.html#intro
>>>
>>> Special thanks to those people on the list who take special care to
>>> ask *smart* questions and those who provide valuable answers. You know
>>> who you are ;)
>>>
>>> +1 - this is actually very handy
>> -MC
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From vbvbrj at gmail.com  Fri Aug 31 19:40:48 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Fri, 31 Aug 2012 18:40:48 +0300
Subject: [Freeswitch-users] Asking a Question (a.k.a. Gabe must be
 getting old)
In-Reply-To: <CAKzWOxXnNQWFwdMTujLUYX6z6yk8x0XuCVSLc=kNnQgkcLcJqQ@mail.gmail.com>
References: <CAMwZ-tJ7Asfd_MSWqNjCdEsSF=DOx9daGZ_T5CbWzoxfTb2UfA@mail.gmail.com>
	<CAKzWOxVKP81wj=86g4_gvRp=d2SkTwA8X_QF5hY2eruTz+iLvg@mail.gmail.com>
	<CAPcNkdfPenKFWF_yauwj6-XpiqJ_PjpOMCA_qN09cdHkZZyJdg@mail.gmail.com>
	<CAKzWOxXnNQWFwdMTujLUYX6z6yk8x0XuCVSLc=kNnQgkcLcJqQ@mail.gmail.com>
Message-ID: <5040DB00.10509@gmail.com>

On 31.08.2012 18:33, Michael Collins wrote:
> FYI, I added a quickie link to the front page for the page Gabe
> suggested. I think this thread has given us our topic for next week's
> conference call: documenting how to find stuff. Let's talk about how we
> find stuff and then write up a nice page that we can link to when people
> ask vague/broad/I'm-really-lazy questions.
>
> -MC

Michael, in mailing list there is lack of sorting by the same 
topic/thread. For example finding thru google a page from mailing list, 
I can't browse all the posts from that page related to thread in question.

-- 
Mimiko desu.


From ntomer at newgen.co.in  Fri Aug 31 19:48:13 2012
From: ntomer at newgen.co.in (Nitin Tomer)
Date: Fri, 31 Aug 2012 21:18:13 +0530
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded
	to	caller itself
In-Reply-To: <CALvF6vAoMBhg0N_jhiEeUFRxHdB2Saiv5iPf=uFNBmWDvxR5XQ@mail.gmail.com>
References: <CALvF6vAoMBhg0N_jhiEeUFRxHdB2Saiv5iPf=uFNBmWDvxR5XQ@mail.gmail.com>
Message-ID: <01ed01cd8790$0800e850$1802b8f0$@co.in>

Thanks Jayesh, it did the trick J

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jayesh
Nambiar
Sent: Friday, August 31, 2012 8:56 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_callcenter: Call being forwarded to
caller itself

 

Nitin,

It is quite possible that you must have added it sometime earlier and only
deleted it from the XML and not from the freeswitch memory. Freeswitch
callcenter module basically maintains an SQLite DB where it stores all these
values. You will have to remove it from the freeswitch memory. Execute this
from freeswitch CLI:

callcenter_config agent del <agent_name>

 

--- Jayesh

 

You are right Jayesh, agent 1000 s being listed from CLI. But it is not in
callcenter.cong.xml. Neither I have added it from CLI. Where it is coming
from?

 

Thanks

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jayesh
Nambiar
Sent: Friday, August 31, 2012 8:07 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_callcenter: Call being forwarded to
caller itself

 

Hi Nitin,

In that case all you have to confirm is that you have not set the contact of
any agent as 1000. Keep in mind that the agents can be pushed into
freeswitch memory from API commands also. Basically you can use FS_CLI to
add agents and set their contacts even without mentioning those agents in
the XML file. So just be sure that you do not have any agent's contact set
to 1000 and also check from FS_CLI issuing this command: "callcenter_config
agent list" to make sure that none of the agent has contact set to 1000.

 

--- Jayesh

 

 

Hi Jayesh,

 

Call is being redirected to call center because it is working other two
options. Even for "sales" the call is redirected to caller(extn 1000) but
when it is declined from there, it works as desired.

 

Thanks

 

Nitin

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jayesh
Nambiar
Sent: Friday, August 31, 2012 6:55 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_callcenter: Call being forwarded to
caller itself

 

Nitin, 

If this is   your exact configuration, it is simply not possible that the
call gets routed back to same extension unless there is something configured
in the dialplan.

Look at the console in loglevel 7 mode and check if the call goes into
callcenter module or no? If no, check the flow of your dialplan and figure
out what is taking the call back to the calling extension.

 

--- Jayesh 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nitin
Tomer

Sent: Wednesday, August 29, 2012 3:39 PM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] mod_callcenter: Call being forwarded to caller
itself

 

Hi,

 

I've configure mod_callcenter on Ubuntu 12.0.4. I've 5 extensions working on
6 soft phones - 1000 to 1005. 1000 is used to call the IVR number, rest 5
are agents.

 

The dial plan looks like -

 

<extension name="ivr_nitin">

      <condition field="destination_number" expression="^5002$">

        <action application="answer"/>

        <action application="sleep" data="500"/>

                <action application="ivr" data="nitin_ivr"/>

      </condition>

    </extension>

<!-- For Call Center -->

<!-- Newgen Support -->

       <extension>

          <condition field="destination_number" expression="^(450)$">

              <action application="set" data="caller_id_name=Newgen Support"
/>

              <action application="set" data="call_timeout=60" />

              <action application="set" data="originate_timeout=60" />

              <action application="callcenter" data="support at default"/>

           </condition>

        </extension>

 

<!-- Newgen Sales -->

        <extension>

           <condition field="destination_number" expression="^(451)$">

               <action application="set" data="caller_id_name=Newgen Sales"
/>

               <action application="set" data="call_timeout=60" />

               <action application="set" data="originate_timeout=60" />

               <action application="callcenter" data="sales at default"/>

           </condition>

        </extension>

 

<!-- Newgen Human Resources -->

        <extension>

           <condition field="destination_number" expression="^(452)$">

               <action application="set" data="caller_id_name=Newgen Human
Resources" />

                       <action application="set" data="call_timeout=60" />

               <action application="set" data="originate_timeout=60" />

               <action application="callcenter" data="hr at default"/>

           </condition>

        </extension>

 

IVR.xml -

 

<include>

        <menu name="nitin_ivr"

                greet-long="say:Welcome to Newgen Software Technologies
Limited. Press 1 for support, 2 for sales or 3 for human resource"

                greet-short="say:Welcome to Newgen. Press 1 for support, 2
for sales or 3 for human resources"

                invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"

                exit-sound="voicemail/vm-goodbye.wav"

                confirm-macro=""

                confirm-key=""

                tts-engine="flite"

                tts-voice="slt"

                confirm-attempts="3"

                timeout="3000"

                inter-digit-timeout="2000"

                max-failures="3"

                max-timeouts="3"

                digit-len="4">

 

                <entry action="menu-exec-app" digits="1" param="transfer 450
XML default"/>    <!-- Support -->

                <entry action="menu-exec-app" digits="2" param="transfer 451
XML default"/>    <!-- Sales -->

                <entry action="menu-exec-app" digits="3" param="transfer 452
XML default"/>    <!-- Human Resources -->

 

        </menu>

</include>

 

And callcenter.conf.xml -

 

<configuration name="callcenter.conf" description="CallCenter">

  <settings>

    <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->

    <!--<param name="dbname" value="/dev/shm/callcenter.db"/>-->

  </settings>

 

    <queues>

                <queue name="sales at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/sales/${strftime(%Y-%m-%d-%H-%M-%S)}.${destin
ation_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="14400"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

                <queue name="support at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${dest
ination_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="60"/>

                                <param name="abandoned-resume-allowed"
value="false"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

                <queue name="hr at default">

                                <param name="strategy"
value="agent-with-least-talk-time"/>

                                <param name="moh-sound"
value="$${hold_music}"/>

      <!--<param name="record-template"
value="$${base_dir}/recordings/support/${strftime(%Y-%m-%d-%H-%M-%S)}.${dest
ination_number}.${caller_id_number}.${uuid}.wav"/>-->

                                <param name="time-base-score"
value="queue"/>

                                <param name="tier-rules-apply"
value="false"/>

                                <param name="tier-rule-wait-second"
value="300"/>

                                <param name="tier-rule-wait-multiply-level"
value="true"/>

                                <param name="tier-rule-no-agent-no-wait"
value="false"/>

                                <param name="discard-abandoned-after"
value="60"/>

                                <param name="abandoned-resume-allowed"
value="false"/>

                                <param name="max-wait-time" value="0"/>

                                <param name="max-wait-time-with-no-agent"
value="120"/>

                </queue>

  </queues>

 

<!-- WARNING: Configuration of XML Agents will be updated into the DB upon
restart. -->

<!-- WARNING: Configuration of XML Tiers will reset the level and position
if those were supplied. -->

<!-- WARNING: Agents and Tiers XML config shouldn't be used in a multi FS
shared DB setup (Not currently supported anyway) -->

  <agents>

    <agent name="1001" type="callback" contact="[call_timeout=10]user/1001"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1002" type="callback" contact="[call_timeout=10]user/1002"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1003" type="callback" contact="[call_timeout=10]user/1003"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1004" type="callback" contact="[call_timeout=10]user/1004"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

     <agent name="1005" type="callback" contact="[call_timeout=10]user/1005"
status="Available" max-no-answer="3" wrap-up-time="10"
reject-delay-time="10" busy-delay-time="60" />

  </agents>

  <tiers>

    <!-- If no level or position is provided, they will default to 1.  You
should do this to keep db value on restart. -->

                <!-- agent 1001 is in all three queues -->

                <tier agent="1001" queue="sales at default" level="1"
position="1"/>

                <tier agent="1001" queue="support at default" level="1"
position="1"/>

                <tier agent="1001" queue="hr at default" level="1"
position="1"/>

 

                <tier agent="1002" queue="sales at default" level="1"
position="1"/>

 

                <tier agent="1003" queue="support at default" level="1"
position="1"/>

 

                <tier agent="1004" queue="sales at default" level="1"
position="1"/>

                <tier agent="1004" queue="support at default" level="1"
position="1"/>

 

                <tier agent="1005" queue="hr at default" level="1"
position="1"/>

  </tiers>

 

</configuration>

 

When I call from extension 1000 and press 2 for sales, call is routed back
to extension 1000, which is not even in agents' list. Why is it happening?

 

Regards

 

Nitin

 

 


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intended recipient and have erroneously received this message, you are
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damages from virus and further acknowledges that no binding nature of the
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due authority of NSTL.



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From freeswitch-users at vocalspace.com  Fri Aug 31 20:57:20 2012
From: freeswitch-users at vocalspace.com (Phillip Boles)
Date: Fri, 31 Aug 2012 11:57:20 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging in
	CS_SOFT_EXEC state
In-Reply-To: <20120830222200.ff3a5e72@mail.tritonwest.net>
References: <20120830222200.ff3a5e72@mail.tritonwest.net>
Message-ID: <54708D96-A6CF-41E4-ADFA-C0AE2BB3BAB9@vocalspace.com>

var session =  ManagedSession.OriginateHandleHangup(s, dialString, ts, (y) => {
	try {
		Log.WriteLine(LogLevel.Info, "Hanging UP: "+ y.GetUuid());
		y.SetAutoHangup(true); 
		y.flushDigits();
		y.flushEvents();
		y.destroy();
		y.Dispose();
		GC.Collect();
	}  catch( Exception ) {
		Log.WriteLine(LogLevel.Critical, "Exception While Trying to handup");	
	}
});
Changes yield no fix. Neither .Dispose() or .destroy() separately or together destroy the channel. I see in the log  the hangup 11da29f3-2d9e-4b74-a439-a96ba60f2db1 but  this is what I get from show channels.
The last log lines of the debug is: 
2012-08-31 11:25:52.109393 [DEBUG] switch_core_state_machine.c:407 (sofia/external/XXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY
2012-08-31 11:25:52.109393 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/XXXXXXXXXX [BREAK]
2012-08-31 11:25:52.109393 [DEBUG] switch_core_session.c:1424 Session 1 (sofia/external/XXXXXXXXXX) Locked, Waiting on external entities


freeswitch at fs03.int.colo> show channels
11da29f3-2d9e-4b74-a439-a96ba60f2db1,outbound,2012-08-31 11:25:24,1346430324,sofia/external/12146635351,CS_SOFT_EXECUTE,Vocalspace,2223334444,,12146635351,,,,default,PCMU,8000,64000,PCMU,8000,64000,,fs03.int.colo,,,ACTIVE,Outbound Call,12146635351,,,,

freeswitch at fs03.int.colo> uuid_kill 11da29f3-2d9e-4b74-a439-a96ba60f2db1

-ERR No Such Channel!

I am calling this from "managed CustomModule.Api"

Calling GC.Collect() later in the execution does not resolve either.
//------------------------------------------------------
// Entrypoint for blocking API execution
//------------------------------------------------------
public void Execute (ApiContext context) {
	
        context.Arguments, context.Event == null ? "<none>" : context.Event.GetEventType ()));

	// this contains the above code
	Run(ParseArguments(context.Arguments));
	GC.Collect();
}

Thanks!
Suggestions appreciated.
On Aug 30, 2012, at 5:22 PM, Dave R. Kompel wrote:

> Actually, all the managed objects are derived from IDisposable, so you should use the .Dispose() method, and let the wrapper do it's job.
> 
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> Sent: Thu, 30 Aug 2012 13:48:07 -0700
> Subject: Re: [Freeswitch-users] Problem with originated calls hanging in CS_SOFT_EXEC state
> 
> destroy method should have a log line about (destroy/unlink session from object)
> try calling your garbage collector, this is common issue with scripts
> and make sure you are on latest GIT build
> 
> 
> On Thu, Aug 30, 2012 at 3:24 PM, Phillip Boles
> <freeswitch-users at vocalspace.com> wrote:
> > Sorry for the excessive logs. Here is my call to originate.
> >
> > var session = ManagedSession.OriginateHandleHangup(s, dialString, ts, (y)
> > => {
> > try {
> > Log.WriteLine(LogLevel.Info, "Hanging UP: "+ y.GetUuid());
> > y.SetAutoHangup(true);
> > y.destroy();
> >
> > } catch( Exception ) {
> > Log.WriteLine(LogLevel.Critical, "Exception While Trying to handup"); 
> > }
> > });
> >
> >
> > My hangup callback is getting hit and I am destroying the session
> >
> > 2012-08-28 10:49:27.296108 [INFO] switch_cpp.cpp:1227 Handing UP:
> > e315f2e8-1fa8-4fd9-849b-f687dad8aed5
> >
> > This is the only call on the system as it is a develpment machine and I see
> > the call state being changed.
> >
> > 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:79
> > sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
> > 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682
> > (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep
> > 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:407
> > (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY
> >
> >
> > If I call show channels after the above output it show there is a session
> > sitting in CS_SOFT_EXEC corresponding to UUID
> > e315f2e8-1fa8-4fd9-849b-f687dad8aed5.
> > Is there something else I need to do to release the lock on this session to
> > let the resources be reclaimed.
> >
> > Thanks!
> >
> > Phillip
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 Aug 31 21:26:39 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 31 Aug 2012 12:26:39 -0500
Subject: [Freeswitch-users] Problem with originated calls hanging in
 CS_SOFT_EXEC state
In-Reply-To: <54708D96-A6CF-41E4-ADFA-C0AE2BB3BAB9@vocalspace.com>
References: <20120830222200.ff3a5e72@mail.tritonwest.net>
	<54708D96-A6CF-41E4-ADFA-C0AE2BB3BAB9@vocalspace.com>
Message-ID: <CAKbxfG9oiNF7NJ==6q1qYWKNq9iq2+zOUboSdX5MmvO712atAQ@mail.gmail.com>

1) You did not answer the question if you are on latest GIT HEAD.  If
you are on anything else update...
2) Add some debugging to switch_cpp.cpp about line 1000

use lines like this to follow the code paths when you call destroy
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "BLAH\n");

The part I am concerned with is when you call destroy you dont see the
log line you should:

switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG,
                              "%s destroy/unlink session from
object\n", switch_channel_get_name(channel));

This makes me wonder if you are some older version...






On Fri, Aug 31, 2012 at 11:57 AM, Phillip Boles
<freeswitch-users at vocalspace.com> wrote:
> var session =  ManagedSession.OriginateHandleHangup(s, dialString, ts, (y)
> => {
> try {
> Log.WriteLine(LogLevel.Info, "Hanging UP: "+ y.GetUuid());
> y.SetAutoHangup(true);
> y.flushDigits();
> y.flushEvents();
> y.destroy();
> y.Dispose();
> GC.Collect();
> }  catch( Exception ) {
> Log.WriteLine(LogLevel.Critical, "Exception While Trying to handup");
> }
> });
> Changes yield no fix. Neither .Dispose() or .destroy() separately or
> together destroy the channel. I see in the log  the hangup
> 11da29f3-2d9e-4b74-a439-a96ba60f2db1 but  this is what I get from show
> channels.
> The last log lines of the debug is:
> 2012-08-31 11:25:52.109393 [DEBUG] switch_core_state_machine.c:407
> (sofia/external/XXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY
> 2012-08-31 11:25:52.109393 [DEBUG] switch_core_session.c:1224 Send signal
> sofia/external/XXXXXXXXXX [BREAK]
> 2012-08-31 11:25:52.109393 [DEBUG] switch_core_session.c:1424 Session 1
> (sofia/external/XXXXXXXXXX) Locked, Waiting on external entities
>
>
> freeswitch at fs03.int.colo> show channels
> 11da29f3-2d9e-4b74-a439-a96ba60f2db1,outbound,2012-08-31
> 11:25:24,1346430324,sofia/external/12146635351,CS_SOFT_EXECUTE,Vocalspace,2223334444,,12146635351,,,,default,PCMU,8000,64000,PCMU,8000,64000,,fs03.int.colo,,,ACTIVE,Outbound
> Call,12146635351,,,,
>
> freeswitch at fs03.int.colo> uuid_kill 11da29f3-2d9e-4b74-a439-a96ba60f2db1
>
> -ERR No Such Channel!
>
> I am calling this from "managed CustomModule.Api"
>
> Calling GC.Collect() later in the execution does not resolve either.
> //------------------------------------------------------
> // Entrypoint for blocking API execution
> //------------------------------------------------------
> public void Execute (ApiContext context) {
>         context.Arguments, context.Event == null ? "<none>" :
> context.Event.GetEventType ()));
>
> // this contains the above code
> Run(ParseArguments(context.Arguments));
> GC.Collect();
> }
>
> Thanks!
> Suggestions appreciated.
> On Aug 30, 2012, at 5:22 PM, Dave R. Kompel wrote:
>
> Actually, all the managed objects are derived from IDisposable, so you
> should use the .Dispose() method, and let the wrapper do it's job.
>
> ________________________________
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> Sent: Thu, 30 Aug 2012 13:48:07 -0700
> Subject: Re: [Freeswitch-users] Problem with originated calls hanging in
> CS_SOFT_EXEC state
>
> destroy method should have a log line about (destroy/unlink session from
> object)
> try calling your garbage collector, this is common issue with scripts
> and make sure you are on latest GIT build
>
>
> On Thu, Aug 30, 2012 at 3:24 PM, Phillip Boles
> <freeswitch-users at vocalspace.com> wrote:
>> Sorry for the excessive logs. Here is my call to originate.
>>
>> var session = ManagedSession.OriginateHandleHangup(s, dialString, ts, (y)
>> => {
>> try {
>> Log.WriteLine(LogLevel.Info, "Hanging UP: "+ y.GetUuid());
>> y.SetAutoHangup(true);
>> y.destroy();
>>
>> } catch( Exception ) {
>> Log.WriteLine(LogLevel.Critical, "Exception While Trying to handup");
>> }
>> });
>>
>>
>> My hangup callback is getting hit and I am destroying the session
>>
>> 2012-08-28 10:49:27.296108 [INFO] switch_cpp.cpp:1227 Handing UP:
>> e315f2e8-1fa8-4fd9-849b-f687dad8aed5
>>
>> This is the only call on the system as it is a develpment machine and I
>> see
>> the call state being changed.
>>
>> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:79
>> sofia/external/XXXXXXXXXXX Standard REPORTING, cause: NORMAL_CLEARING
>> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:682
>> (sofia/external/XXXXXXXXXXX) State REPORTING going to sleep
>> 2012-08-28 10:49:27.296108 [DEBUG] switch_core_state_machine.c:407
>> (sofia/external/XXXXXXXXXXX) State Change CS_REPORTING -> CS_DESTROY
>>
>>
>> If I call show channels after the above output it show there is a session
>> sitting in CS_SOFT_EXEC corresponding to UUID
>> e315f2e8-1fa8-4fd9-849b-f687dad8aed5.
>> Is there something else I need to do to release the lock on this session
>> to
>> let the resources be reclaimed.
>>
>> Thanks!
>>
>> Phillip
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

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

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


From lon at kickasspixels.com  Fri Aug 31 22:17:05 2012
From: lon at kickasspixels.com (Lon Baker)
Date: Fri, 31 Aug 2012 11:17:05 -0700
Subject: [Freeswitch-users] Eavesdrop with bind_digit_action
In-Reply-To: <mailman.17974.1346354118.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.17974.1346354118.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <E81B9B3D-22FC-4DC5-912D-F5C319E424CB@kickasspixels.com>

Michael - Oops, Valid point.

I've tried numerous techniques and discovered the issue, I think.

bind_digit_action doesn't run inline. Instead it queues the action, which doesn't do anything while the channel is in eavesdrop. The bind_meta_app has the option of running inline using the "i" flag, which appears to function as desired.

The other difference is that I can designate any DTMF combination for bind_digit_action. While bind_meta_app only accepts a metakey + single digit. The desired functionality I'm looking for was to use a single digit without metakey. 


Dialplan for bind_digit_action (FAILS):

> <action application="bind_digit_action" data="my_digits,1,exec:execute_extension,pull_from_voicemail XML default"/>
> <action application="digit_action_set_realm" data="my_digits"/>
> <action application="eavesdrop" data="${originating_leg_uuid}"/>


Results:

> RTP RECV DTMF 1:1600
> Digit match binding [exec:execute_extension][pull_from_voicemail XML default]


-------

Dialplan for bind_meta_app (WORKS):

> <action application="bind_meta_app" data="1 a i execute_extension::pull_from_voicemail XML det"/>
> <action application="eavesdrop" data="${originating_leg_uuid}"/>


Results:

> RTP RECV DTMF #:1600
> RTP RECV DTMF 1:1600
> Processing meta digit '1' [execute_extension::pull_from_voicemail XML det]
> EXECUTE sofia/internal/sip:1001 at 10.0.1.7:60446 execute_extension(pull_from_voicemail XML det)
> Processing Outbound Call <1001>->pull_from_voicemail in context det

--
Lon

On Aug 30, 2012, at 12:15 PM, freeswitch-users-request at lists.freeswitch.org wrote:

> Malfunction! Need input! Can you throw some logs and config info on
> pastebin? I think some folks might be interested in seeing this problem
> solved and if you've already started the work I bet others would be willing
> to assist.

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From msc at freeswitch.org  Fri Aug 31 22:22:04 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 31 Aug 2012 11:22:04 -0700
Subject: [Freeswitch-users] Asking a Question (a.k.a. Gabe must be
	getting old)
In-Reply-To: <5040DB00.10509@gmail.com>
References: <CAMwZ-tJ7Asfd_MSWqNjCdEsSF=DOx9daGZ_T5CbWzoxfTb2UfA@mail.gmail.com>
	<CAKzWOxVKP81wj=86g4_gvRp=d2SkTwA8X_QF5hY2eruTz+iLvg@mail.gmail.com>
	<CAPcNkdfPenKFWF_yauwj6-XpiqJ_PjpOMCA_qN09cdHkZZyJdg@mail.gmail.com>
	<CAKzWOxXnNQWFwdMTujLUYX6z6yk8x0XuCVSLc=kNnQgkcLcJqQ@mail.gmail.com>
	<5040DB00.10509@gmail.com>
Message-ID: <CAKzWOxU1m99bo1Z=Ysduw_hXVT_Wrk23b0wRNVFVN934-8rHSg@mail.gmail.com>

On Fri, Aug 31, 2012 at 8:40 AM, Vbvbrj <vbvbrj at gmail.com> wrote:

> On 31.08.2012 18:33, Michael Collins wrote:
> > FYI, I added a quickie link to the front page for the page Gabe
> > suggested. I think this thread has given us our topic for next week's
> > conference call: documenting how to find stuff. Let's talk about how we
> > find stuff and then write up a nice page that we can link to when people
> > ask vague/broad/I'm-really-lazy questions.
> >
> > -MC
>
> Michael, in mailing list there is lack of sorting by the same
> topic/thread. For example finding thru google a page from mailing list,
> I can't browse all the posts from that page related to thread in question.
>
> You can at lists.freeswitch.org. For example, this thread is found here:
http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/087411.html
-MC


> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From vbvbrj at gmail.com  Fri Aug 31 22:45:50 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Fri, 31 Aug 2012 21:45:50 +0300
Subject: [Freeswitch-users] Asking a Question (a.k.a. Gabe must be
 getting old)
In-Reply-To: <CAKzWOxU1m99bo1Z=Ysduw_hXVT_Wrk23b0wRNVFVN934-8rHSg@mail.gmail.com>
References: <CAMwZ-tJ7Asfd_MSWqNjCdEsSF=DOx9daGZ_T5CbWzoxfTb2UfA@mail.gmail.com>
	<CAKzWOxVKP81wj=86g4_gvRp=d2SkTwA8X_QF5hY2eruTz+iLvg@mail.gmail.com>
	<CAPcNkdfPenKFWF_yauwj6-XpiqJ_PjpOMCA_qN09cdHkZZyJdg@mail.gmail.com>
	<CAKzWOxXnNQWFwdMTujLUYX6z6yk8x0XuCVSLc=kNnQgkcLcJqQ@mail.gmail.com>
	<5040DB00.10509@gmail.com>
	<CAKzWOxU1m99bo1Z=Ysduw_hXVT_Wrk23b0wRNVFVN934-8rHSg@mail.gmail.com>
Message-ID: <5041065E.1050302@gmail.com>

On 31.08.2012 21:22, Michael Collins wrote:
> You can at lists.freeswitch.org <http://lists.freeswitch.org>. For
> example, this thread is found here:
> http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/087411.html
> -MC

Well, if you see the links bellow:
Previous message: [Freeswitch-users] No Dialplan on answer channel
Next message: [Freeswitch-users] Asking a Question (a.k.a. Gabe must be 
getting old)

So one cant browse back to previous messages for thise thread unless 
step other other threads.

-- 
Mimiko desu.


From msc at freeswitch.org  Fri Aug 31 23:14:40 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 31 Aug 2012 12:14:40 -0700
Subject: [Freeswitch-users] Asking a Question (a.k.a. Gabe must be
	getting old)
In-Reply-To: <5041065E.1050302@gmail.com>
References: <CAMwZ-tJ7Asfd_MSWqNjCdEsSF=DOx9daGZ_T5CbWzoxfTb2UfA@mail.gmail.com>
	<CAKzWOxVKP81wj=86g4_gvRp=d2SkTwA8X_QF5hY2eruTz+iLvg@mail.gmail.com>
	<CAPcNkdfPenKFWF_yauwj6-XpiqJ_PjpOMCA_qN09cdHkZZyJdg@mail.gmail.com>
	<CAKzWOxXnNQWFwdMTujLUYX6z6yk8x0XuCVSLc=kNnQgkcLcJqQ@mail.gmail.com>
	<5040DB00.10509@gmail.com>
	<CAKzWOxU1m99bo1Z=Ysduw_hXVT_Wrk23b0wRNVFVN934-8rHSg@mail.gmail.com>
	<5041065E.1050302@gmail.com>
Message-ID: <CAKzWOxVxyjC6bdnLqxcy+SuM9zJwUb7jg3=Uji2RnuCZxcGG0g@mail.gmail.com>

I'm afraid that this is nitpicking. You can view any post in any thread.
This is Mailman for crying out loud! Everyone uses it - we are no better or
worse than the thousands of other mailing lists that have it. If you are
complaining because it takes two clicks instead of one then I'm afraid you
won't find an receptive audience around here...

-MC

On Fri, Aug 31, 2012 at 11:45 AM, Vbvbrj <vbvbrj at gmail.com> wrote:

> On 31.08.2012 21:22, Michael Collins wrote:
> > You can at lists.freeswitch.org <http://lists.freeswitch.org>. For
> > example, this thread is found here:
> >
> http://lists.freeswitch.org/pipermail/freeswitch-users/2012-August/087411.html
> > -MC
>
> Well, if you see the links bellow:
> Previous message: [Freeswitch-users] No Dialplan on answer channel
> Next message: [Freeswitch-users] Asking a Question (a.k.a. Gabe must be
> getting old)
>
> So one cant browse back to previous messages for thise thread unless
> step other other threads.
>
> --
> Mimiko desu.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From vbvbrj at gmail.com  Fri Aug 31 23:19:04 2012
From: vbvbrj at gmail.com (Vbvbrj)
Date: Fri, 31 Aug 2012 22:19:04 +0300
Subject: [Freeswitch-users] Asking a Question (a.k.a. Gabe must be
 getting old)
In-Reply-To: <CAKzWOxVxyjC6bdnLqxcy+SuM9zJwUb7jg3=Uji2RnuCZxcGG0g@mail.gmail.com>
References: <CAMwZ-tJ7Asfd_MSWqNjCdEsSF=DOx9daGZ_T5CbWzoxfTb2UfA@mail.gmail.com>
	<CAKzWOxVKP81wj=86g4_gvRp=d2SkTwA8X_QF5hY2eruTz+iLvg@mail.gmail.com>
	<CAPcNkdfPenKFWF_yauwj6-XpiqJ_PjpOMCA_qN09cdHkZZyJdg@mail.gmail.com>
	<CAKzWOxXnNQWFwdMTujLUYX6z6yk8x0XuCVSLc=kNnQgkcLcJqQ@mail.gmail.com>
	<5040DB00.10509@gmail.com>
	<CAKzWOxU1m99bo1Z=Ysduw_hXVT_Wrk23b0wRNVFVN934-8rHSg@mail.gmail.com>
	<5041065E.1050302@gmail.com>
	<CAKzWOxVxyjC6bdnLqxcy+SuM9zJwUb7jg3=Uji2RnuCZxcGG0g@mail.gmail.com>
Message-ID: <50410E28.50904@gmail.com>

On 31.08.2012 22:14, Michael Collins wrote:
> I'm afraid that this is nitpicking. You can view any post in any thread.
> This is Mailman for crying out loud! Everyone uses it - we are no better
> or worse than the thousands of other mailing lists that have it. If you
> are complaining because it takes two clicks instead of one then I'm
> afraid you won't find an receptive audience around here...
>
> -MC

There is no complain, I know about this is all mailing lists behavior. 
Its just a suggestion if it is possible.

-- 
Mimiko desu.


From mike.burlingame at me.com  Fri Aug 31 23:33:03 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 31 Aug 2012 12:33:03 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <CAJUJwthhd8jj4ZGeFnFv-oRS8Xvst_T+a10_ecbomrwG58-LXw@mail.gmail.com>
References: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
	<8D066FE8-DB24-4928-8508-317E23B858EA@me.com>
	<CAJUJwthxmhd0u6G35ANgiCzdHv1KY-qKK5ZD_Xx_+S-cGYMXVg@mail.gmail.com>
	<9065DE2C-DF4A-4E6C-B436-1A83B2AFA664@me.com>
	<CAJUJwthU62H9o3iutVpt3EZO2-UKW0PXk6fJgFeGUO7NT1W0VQ@mail.gmail.com>
	<AD70ED57-CDBB-4F9D-AE0F-77008F9EB45E@me.com>
	<CAJUJwthhd8jj4ZGeFnFv-oRS8Xvst_T+a10_ecbomrwG58-LXw@mail.gmail.com>
Message-ID: <C9A00194-61D5-4DB8-A5F6-0953F071CEAF@me.com>

I do have an OpenSIPs proxy in front of and behind the A and B Leg's - but the issue still seems to be in FS

Sonus_UAC --> OpenSIPs --> Freeswitch --> OpenSIPs --> Broadsoft_M6

Log and SIP trace from FS Log file posted to PasteBin: http://pastebin.freeswitch.org/19799


On Aug 31, 2012, at 12:14 AM, SamyGo <govoiper at gmail.com> wrote:

> Well that narrow downs the possible solutions to doing some stuff on FS only ! - I really think that Stopping or applying a work-around for the re-invite from Broadsoft will get this fixed. I did some googling on the SIP 491 code and re-invites in FS and mostly they point to session timers or some media_bridging option in dialplan.
> 
> 
> On Fri, Aug 31, 2012 at 12:03 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> The Sonus and M6 are out of my control they belong to my upstream and downstream so I have no access to them
> 
> Broadsoft support will not work this issue because FS ACKs the 200 OK so dialog is completed at least on the B-Leg so that's when the M6 sends the 2nd invite however the dialog has not completed on the A-Leg but FS still passes the Invite along resulting in the 491 
> 
> 
> Sent from my iPhone 4S
> 
> On Aug 30, 2012, at 11:42 PM, SamyGo <govoiper at gmail.com> wrote:
> 
>> OK, and can you see whats happening inside the SONUS when it gets the second invite (debug logs etc)? Can you control the M6 to stop re-invites !?
>> 
>> On Fri, Aug 31, 2012 at 10:29 AM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> Sure I will put logs and sip dump up tomorrow A-Leg == Sonus B-Leg == Broadsoft M6
>> 
>> Sent from my iPhone 4S
>> 
>> On Aug 30, 2012, at 10:24 PM, SamyGo <govoiper at gmail.com> wrote:
>> 
>>> Hi,
>>> Can you please manage a proper sip trace for this ! I'd really love to troubleshoot and analyse the sip traces.
>>> What is the equipment on your A-leg ? 
>>> 
>>> BR
>>> Sammy
>>> 
>>>  
>>> 
>>> On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> I would like to add an update to this and maybe someone has some other suggestions - It does look like that FS is disconnecting the call with cause code 31  [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on the Bye when I first looked at the log's
>>> 
>>> The basis of the issue still stands a 491 Results in the B-Leg being disconnected
>>> 
>>> On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>>> 
>>>> We are seeing some instances when we send a invite from the B-Leg back to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has not fully completed yet causing a return of a 491 from the A-Leg side causing the call to be disconnected. wanted to see if anyone else has seen something like this while running FS and if anyone had any suggestions on a fix?
>>>> 
>>>> A-Leg
>>>> Invite into Freeswitch
>>>> 100 Trying back from FS to A-Leg
>>>> 180 Ringing from FS to A-Leg
>>>> 200 OK from FS to A-Leg at 15:08:14.638799
>>>> Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515
>>>> 100 Giving a try form A-Leg to FS at 15:08:14.749757
>>>> 491 From A-Leg to FS at 15:08:14.780968
>>>> ACK from FS to A-Leg at 15:08:14.781102
>>>> ACK from FS to A-Leg at 15:08:14.797143
>>>> BYE from A-LEG to FS 15:11:10.791963
>>>    481 Call Does Not Exist back to A-LEG
>>>> 
>>>> B-Leg
>>>> Invite from Freeswitch to B-Leg
>>>> 100 Giving a try from B-Leg
>>>> 180 Ringing from B-Leg
>>>> 200 OK  from B-Leg at 15:08:14.635670
>>>> ACK from FS to B-Leg  at 15:08:14.637044
>>>> Invite from B-Leg to FS at 15:08:14.748623
>>>> 100 Trying from FS to B-Leg at 15:08:14.748954
>>>> 491 from FS to B-Leg at 15:08:14.782169
>>>> ACK from B-Leg to FS at 15:08:14.782372
>>>> BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 cesar.bermudez at gmail.com  Fri Aug 31 23:40:48 2012
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Fri, 31 Aug 2012 13:40:48 -0600
Subject: [Freeswitch-users]  0800 providers?
Message-ID: <CADrcvV-dQLeyfYqc5Dt_Y0JM_o6YeMLLvEMjFwiNBVNx1TtO-w@mail.gmail.com>

Hi guys, what provider can give me termination for my users can dial 0800
to USA.
Best regards.
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From anthony.minessale at gmail.com  Fri Aug 31 23:50:08 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 31 Aug 2012 14:50:08 -0500
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <C9A00194-61D5-4DB8-A5F6-0953F071CEAF@me.com>
References: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
	<8D066FE8-DB24-4928-8508-317E23B858EA@me.com>
	<CAJUJwthxmhd0u6G35ANgiCzdHv1KY-qKK5ZD_Xx_+S-cGYMXVg@mail.gmail.com>
	<9065DE2C-DF4A-4E6C-B436-1A83B2AFA664@me.com>
	<CAJUJwthU62H9o3iutVpt3EZO2-UKW0PXk6fJgFeGUO7NT1W0VQ@mail.gmail.com>
	<AD70ED57-CDBB-4F9D-AE0F-77008F9EB45E@me.com>
	<CAJUJwthhd8jj4ZGeFnFv-oRS8Xvst_T+a10_ecbomrwG58-LXw@mail.gmail.com>
	<C9A00194-61D5-4DB8-A5F6-0953F071CEAF@me.com>
Message-ID: <CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>

An idea I had you can test if you update to latest

set the variable sip_wait_for_aleg_ack on the b leg in the {} when you
make the outbound call

<action application="bridge"
data="{sip_wait_for_aleg_ack=true}sofia/internal/foo at bar.com"/>

In theory this should make the B leg delay sending the ACK until it
sees that the A leg has recv'd an ack



On Fri, Aug 31, 2012 at 2:33 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
> I do have an OpenSIPs proxy in front of and behind the A and B Leg's - but
> the issue still seems to be in FS
>
> Sonus_UAC --> OpenSIPs --> Freeswitch --> OpenSIPs --> Broadsoft_M6
>
> Log and SIP trace from FS Log file posted to PasteBin:
> http://pastebin.freeswitch.org/19799
>
>
> On Aug 31, 2012, at 12:14 AM, SamyGo <govoiper at gmail.com> wrote:
>
> Well that narrow downs the possible solutions to doing some stuff on FS only
> ! - I really think that Stopping or applying a work-around for the re-invite
> from Broadsoft will get this fixed. I did some googling on the SIP 491 code
> and re-invites in FS and mostly they point to session timers or some
> media_bridging option in dialplan.
>
>
> On Fri, Aug 31, 2012 at 12:03 PM, Mike Burlingame <mike.burlingame at me.com>
> wrote:
>>
>> The Sonus and M6 are out of my control they belong to my upstream and
>> downstream so I have no access to them
>>
>> Broadsoft support will not work this issue because FS ACKs the 200 OK so
>> dialog is completed at least on the B-Leg so that's when the M6 sends the
>> 2nd invite however the dialog has not completed on the A-Leg but FS still
>> passes the Invite along resulting in the 491
>>
>>
>> Sent from my iPhone 4S
>>
>> On Aug 30, 2012, at 11:42 PM, SamyGo <govoiper at gmail.com> wrote:
>>
>> OK, and can you see whats happening inside the SONUS when it gets the
>> second invite (debug logs etc)? Can you control the M6 to stop re-invites !?
>>
>> On Fri, Aug 31, 2012 at 10:29 AM, Mike Burlingame <mike.burlingame at me.com>
>> wrote:
>>>
>>> Sure I will put logs and sip dump up tomorrow A-Leg == Sonus B-Leg ==
>>> Broadsoft M6
>>>
>>> Sent from my iPhone 4S
>>>
>>> On Aug 30, 2012, at 10:24 PM, SamyGo <govoiper at gmail.com> wrote:
>>>
>>> Hi,
>>> Can you please manage a proper sip trace for this ! I'd really love to
>>> troubleshoot and analyse the sip traces.
>>> What is the equipment on your A-leg ?
>>>
>>> BR
>>> Sammy
>>>
>>>
>>>
>>> On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame at me.com>
>>> wrote:
>>>>
>>>> I would like to add an update to this and maybe someone has some other
>>>> suggestions - It does look like that FS is disconnecting the call with cause
>>>> code 31  [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on
>>>> the Bye when I first looked at the log's
>>>>
>>>> The basis of the issue still stands a 491 Results in the B-Leg being
>>>> disconnected
>>>>
>>>> On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame at me.com>
>>>> wrote:
>>>>
>>>> We are seeing some instances when we send a invite from the B-Leg back
>>>> to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has
>>>> not fully completed yet causing a return of a 491 from the A-Leg side
>>>> causing the call to be disconnected. wanted to see if anyone else has seen
>>>> something like this while running FS and if anyone had any suggestions on a
>>>> fix?
>>>>
>>>> A-Leg
>>>> Invite into Freeswitch
>>>> 100 Trying back from FS to A-Leg
>>>> 180 Ringing from FS to A-Leg
>>>> 200 OK from FS to A-Leg at 15:08:14.638799
>>>> Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515
>>>> 100 Giving a try form A-Leg to FS at 15:08:14.749757
>>>> 491 From A-Leg to FS at 15:08:14.780968
>>>> ACK from FS to A-Leg at 15:08:14.781102
>>>> ACK from FS to A-Leg at 15:08:14.797143
>>>> BYE from A-LEG to FS 15:11:10.791963
>>>>
>>>>    481 Call Does Not Exist back to A-LEG
>>>>
>>>>
>>>> B-Leg
>>>> Invite from Freeswitch to B-Leg
>>>> 100 Giving a try from B-Leg
>>>> 180 Ringing from B-Leg
>>>> 200 OK  from B-Leg at 15:08:14.635670
>>>> ACK from FS to B-Leg  at 15:08:14.637044
>>>> Invite from B-Leg to FS at 15:08:14.748623
>>>> 100 Trying from FS to B-Leg at 15:08:14.748954
>>>> 491 from FS to B-Leg at 15:08:14.782169
>>>> ACK from B-Leg to FS at 15:08:14.782372
>>>> BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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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
>



-- 
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 mike.burlingame at me.com  Fri Aug 31 23:59:36 2012
From: mike.burlingame at me.com (Mike Burlingame)
Date: Fri, 31 Aug 2012 12:59:36 -0700
Subject: [Freeswitch-users] B-Leg / A-Leg Race Condition
In-Reply-To: <CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
References: <D5988BB4-37BB-4EAD-9E61-3695C33020FE@me.com>
	<8D066FE8-DB24-4928-8508-317E23B858EA@me.com>
	<CAJUJwthxmhd0u6G35ANgiCzdHv1KY-qKK5ZD_Xx_+S-cGYMXVg@mail.gmail.com>
	<9065DE2C-DF4A-4E6C-B436-1A83B2AFA664@me.com>
	<CAJUJwthU62H9o3iutVpt3EZO2-UKW0PXk6fJgFeGUO7NT1W0VQ@mail.gmail.com>
	<AD70ED57-CDBB-4F9D-AE0F-77008F9EB45E@me.com>
	<CAJUJwthhd8jj4ZGeFnFv-oRS8Xvst_T+a10_ecbomrwG58-LXw@mail.gmail.com>
	<C9A00194-61D5-4DB8-A5F6-0953F071CEAF@me.com>
	<CAKbxfG-Twz-XA_rf9sYF4CY5zzxXFVYQaD_JAzmOZVU4bBQzxw@mail.gmail.com>
Message-ID: <5F7F4610-2849-4141-AC12-B8872C0303FC@me.com>

Cool I will nail that up on my test box and see if that works

Thanks for the suggestion

Sent from my iPhone 4S

On Aug 31, 2012, at 12:50 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> An idea I had you can test if you update to latest
> 
> set the variable sip_wait_for_aleg_ack on the b leg in the {} when you
> make the outbound call
> 
> <action application="bridge"
> data="{sip_wait_for_aleg_ack=true}sofia/internal/foo at bar.com"/>
> 
> In theory this should make the B leg delay sending the ACK until it
> sees that the A leg has recv'd an ack
> 
> 
> 
> On Fri, Aug 31, 2012 at 2:33 PM, Mike Burlingame <mike.burlingame at me.com> wrote:
>> I do have an OpenSIPs proxy in front of and behind the A and B Leg's - but
>> the issue still seems to be in FS
>> 
>> Sonus_UAC --> OpenSIPs --> Freeswitch --> OpenSIPs --> Broadsoft_M6
>> 
>> Log and SIP trace from FS Log file posted to PasteBin:
>> http://pastebin.freeswitch.org/19799
>> 
>> 
>> On Aug 31, 2012, at 12:14 AM, SamyGo <govoiper at gmail.com> wrote:
>> 
>> Well that narrow downs the possible solutions to doing some stuff on FS only
>> ! - I really think that Stopping or applying a work-around for the re-invite
>> from Broadsoft will get this fixed. I did some googling on the SIP 491 code
>> and re-invites in FS and mostly they point to session timers or some
>> media_bridging option in dialplan.
>> 
>> 
>> On Fri, Aug 31, 2012 at 12:03 PM, Mike Burlingame <mike.burlingame at me.com>
>> wrote:
>>> 
>>> The Sonus and M6 are out of my control they belong to my upstream and
>>> downstream so I have no access to them
>>> 
>>> Broadsoft support will not work this issue because FS ACKs the 200 OK so
>>> dialog is completed at least on the B-Leg so that's when the M6 sends the
>>> 2nd invite however the dialog has not completed on the A-Leg but FS still
>>> passes the Invite along resulting in the 491
>>> 
>>> 
>>> Sent from my iPhone 4S
>>> 
>>> On Aug 30, 2012, at 11:42 PM, SamyGo <govoiper at gmail.com> wrote:
>>> 
>>> OK, and can you see whats happening inside the SONUS when it gets the
>>> second invite (debug logs etc)? Can you control the M6 to stop re-invites !?
>>> 
>>> On Fri, Aug 31, 2012 at 10:29 AM, Mike Burlingame <mike.burlingame at me.com>
>>> wrote:
>>>> 
>>>> Sure I will put logs and sip dump up tomorrow A-Leg == Sonus B-Leg ==
>>>> Broadsoft M6
>>>> 
>>>> Sent from my iPhone 4S
>>>> 
>>>> On Aug 30, 2012, at 10:24 PM, SamyGo <govoiper at gmail.com> wrote:
>>>> 
>>>> Hi,
>>>> Can you please manage a proper sip trace for this ! I'd really love to
>>>> troubleshoot and analyse the sip traces.
>>>> What is the equipment on your A-leg ?
>>>> 
>>>> BR
>>>> Sammy
>>>> 
>>>> 
>>>> 
>>>> On Fri, Aug 31, 2012 at 3:28 AM, Mike Burlingame <mike.burlingame at me.com>
>>>> wrote:
>>>>> 
>>>>> I would like to add an update to this and maybe someone has some other
>>>>> suggestions - It does look like that FS is disconnecting the call with cause
>>>>> code 31  [NORMAL_UNSPECIFIED] I did not notice the time stamp difference on
>>>>> the Bye when I first looked at the log's
>>>>> 
>>>>> The basis of the issue still stands a 491 Results in the B-Leg being
>>>>> disconnected
>>>>> 
>>>>> On Aug 24, 2012, at 12:53 PM, Mike Burlingame <mike.burlingame at me.com>
>>>>> wrote:
>>>>> 
>>>>> We are seeing some instances when we send a invite from the B-Leg back
>>>>> to FS and FS passes the invite to the A-Leg that the dialog on the A-Leg has
>>>>> not fully completed yet causing a return of a 491 from the A-Leg side
>>>>> causing the call to be disconnected. wanted to see if anyone else has seen
>>>>> something like this while running FS and if anyone had any suggestions on a
>>>>> fix?
>>>>> 
>>>>> A-Leg
>>>>> Invite into Freeswitch
>>>>> 100 Trying back from FS to A-Leg
>>>>> 180 Ringing from FS to A-Leg
>>>>> 200 OK from FS to A-Leg at 15:08:14.638799
>>>>> Invite from FS to A-Leg (From B-Leg below) at 15:08:14.749515
>>>>> 100 Giving a try form A-Leg to FS at 15:08:14.749757
>>>>> 491 From A-Leg to FS at 15:08:14.780968
>>>>> ACK from FS to A-Leg at 15:08:14.781102
>>>>> ACK from FS to A-Leg at 15:08:14.797143
>>>>> BYE from A-LEG to FS 15:11:10.791963
>>>>> 
>>>>>   481 Call Does Not Exist back to A-LEG
>>>>> 
>>>>> 
>>>>> B-Leg
>>>>> Invite from Freeswitch to B-Leg
>>>>> 100 Giving a try from B-Leg
>>>>> 180 Ringing from B-Leg
>>>>> 200 OK  from B-Leg at 15:08:14.635670
>>>>> ACK from FS to B-Leg  at 15:08:14.637044
>>>>> Invite from B-Leg to FS at 15:08:14.748623
>>>>> 100 Trying from FS to B-Leg at 15:08:14.748954
>>>>> 491 from FS to B-Leg at 15:08:14.782169
>>>>> ACK from B-Leg to FS at 15:08:14.782372
>>>>> BYE from FS to B-Leg at 15:08:14.790714 with Cause Code 31
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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